innovaphone wiki - User contributions [en]

Howto:DE - Viakom - Voice WB20 TCP SIP-Provider (2026)

2026-08-14T12:04:52Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=August 14th, 2026|url=https://www.viakom.de/produkte/service-providing/professional-service/it-service-support/|productname=Voice_WB20_TCP|providername=Viakom}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_but_clns_onnet_NOK}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Viakom-Voice_WB20_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Reference16r1:Concept App Service Conference Transcriptions

2026-08-14T08:54:42Z

Can: /* Installation */

[[Category:Concept|Apps]]
[[Category:Concept App Service Conference Transcriptions]]
== Applies To ==

* innovaphone Conference Transciptions from version 16r1

== Overview ==
[[File:Reference16r1 Concept App Service Conference Transcriptions Softphone App.png|thumb|309x309px|/Reference16r1_Concept_App_Service_Conference_Transcriptions_Softphone_App.png]]
The main goal of the Conference Transcriptions Service is to create video‑conference meeting summaries, which can be accessed in the Phone Apps.

To achieve this:
# Users can start a transcription of the conference audio by clicking the '''“Transcribe Conference”''' button in the Phone Apps.
#* This action makes the Conference Transcriptions service connect to the conference room.
#* It records the audio and creates audio files of approximately one minute in length.
#* It forwards these small audio files via the com.innovaphone.transcriptions API, offered by the Transcriptions Service ([[{{NAMESPACE}}:Concept_App_Service_Transcriptions | Concept App Service Transcriptions]])
#* It receives the transcripts of these small files and stores them in the service database.
# If the user stops the transcription by clicking on the "'''Transcribe Conference'''" Button again
#* The Conference Transcriptions Service gathers the transcribed text from the database
#* It retrieves the prompt configured in the settings plugin [[{{NAMESPACE}}:Apps/PbxManager/App myApps ConferenceTranscriptions | ConferenceTranscriptions settings plugin]]
#* It sends the transcript together with the prompt to the Assistant App Service via the com.innovaphone.assistant API
#* It receives the meeting summary and stores that summary in the database of the service
# Users may start and stop transcriptions multiple times during a meeting. As long as users remain in the room, the service recognizes that the meeting is still the same for that room.
# After the meeting, users can find the AI‑generated meeting summary in the call list of their Phone Apps by clicking on the '''“i”''' button. 

== Licensing ==
* In order to use the service each user needs the newly introduced '''UCC license'''
* Also one '''Port License''' is used by the app service to establish a VoIP registration at the PBX

== Installation ==
In order to make the service work properly, two supplemental services must be installed first:
* the myApps Assistant which will summarize the transcript [[{{NAMESPACE}}:Concept App Service myApps Assistant | Concept App Service myApps Assistant]]
* the Transcriptions Service which establishes a connection to a remote transcription server [[{{NAMESPACE}}:Concept_App_Service_Transcriptions | Concept App Service Transcriptions]]

Go to the '''Settings App''' (fomerly known as PBX manager) and open the '''"AP app installer"''' plugin. On the right panel, the App Store will be shown. ''Hint : if you access it for the first time, you will need to accept the "Terms of Use of the innovaphone App Store"''
* In the search field located on the top right corner of the store, search for '''"Conference Transcriptions"''' and click on it.
* Select the proper firmware version, for example '''"Version 16r1"''' and click on install.
* Tick "I accept the terms of use" and continue by clicking on the install yellow button.
* Wait until the install has been finished.
* Close and reopen the '''Settings App''' (PBX manager) in order to refresh the list of the available colored AP plugin icons.
* Open the '''"AP Conference Transcriptions"''' Plugin by clicking on it and click on '''" + Add an App"''' and then on the '''"Conference Transcriptions API"''' button.
* Enter a '''"Name"''' ''(used as display name / all character allowed)'' and the '''"SIP"''' ''(no space, no capital letters)''. ''e.g : Name: Conference Transcriptions API, SIP: conference-transcriptions-api''
* Choose a LLM (model) from the dropdown
* Tick the appropriate template to distribute the App (the app is needed at every user object from any user who wants to use the assistant API)
* Click OK to save the settings. A green check mark will be shown to confirm that the configuration is good.

== Conference Transcriptions - App Service ==
The App Service performs tasks in the following areas:
* Transcribes the audio of a conference room
* Stores the transcript in the database
* Creates and stores the summary of the transcript in the database

It can be configured in the '''Settings App''' [[{{NAMESPACE}}:Apps/PbxManager/App myApps ConferenceTranscriptions | ConferenceTranscriptions settings plugin]].

== Phone App Integration ==
Users may control the generated transcripts using the Phone Apps. If the default transcript generated right after the meeting is not adequate users may generate a new customized one using the button "Customize style". They may:
[[File:Concept-Conference-Transcriptions-Softphone-Focus.png|left|thumb|220x220px|Focus|/Concept-Conference-Transcriptions-Softphone-Focus.png]]
[[File:Concept-Conference-Transcriptions-Softphone-Prompts-Choice.png|thumb|215x215px|Choose prompt|/Concept-Conference-Transcriptions-Softphone-Prompts-Choice.png]]

* Choose a different predefined prompt from the drop-down menu (or use no prompt at all choosing the "-")
* Give some extra info or requests using the input field "What should be the focus?"
** This may be something like "Translate the transcript in Korean!"
** or "Just write down which decisions were made!", "There were some links pasted in the chat. Please integrate them in the meeting summary!"

== Summary prompts ==
You may configure the prompts freely. Here are some propositions you may copy/paste into the corresponding field of the settings plugin. Depending on the used model these prompts don't need to be that sophisticated. Something like "Summarize the following transcript! Make clear who has taken which task!" may be sufficient.

'''1. General prompt'''

<pre>
You will receive a meeting transcript. Your task is to turn it into a structured meeting summary.

The entire summary — including section headers, bullet points, and all text — must be written in the same language as the transcript. Detect the language automatically. If the transcript is in german write the summary in german. If the transcript is in english, write the summary in english...

Follow these instructions exactly:

1. Read the transcript carefully.
2. Extract only the information that is clearly stated. Do not guess or invent anything.
3. Create the final summary using the exact structure below, but translate the section headers into the language of the transcript.

OUTPUT FORMAT (translate the headers into the transcript language, but keep the structure):

 
A short title describing the meeting.
 
 

Summary: 
 
(2–4 sentences describing the purpose of the meeting and the main outcomes.)
 
 
Decisions:
 
 
<ul>
<li>Decision 1</li>
<li>Decision 2</li>

</ul>
 
 
Key Points:
 
 
<ul>
<li>Bullet point 1</li>
<li>Bullet point 2</li>
<li>Bullet point 3</li>


</ul>

 
 
Tasks:
 
 
<ul>
<li>Person (with sip e.g. @pso)— Task — Deadline (if mentioned)</li>

</ul>
 
 
Open Questions:
 
 
<ul>
<li>Question 1</li>
<li>Question 2</li>

</ul>

RULES:

Use HTML lists (<ul><li></li></ul>) only for bullet points.
Do not add information that is not in the transcript.
Keep the writing clear and concise.
Keep the structure exactly as shown, but translate the headers.
Do not wrap the whole answer in a single HTML tag.

TRANSCRIPT:
</pre>

'''2. A more structured version:'''

<pre>
You will receive a meeting transcript. Your task is to turn it into a structured meeting summary.

The entire summary — including section headers, bullet points, and all text — must be written in the same language as the transcript. Detect the language automatically. If the transcript is in german write the summary in german. If the transcript is in english, write the summary in english...

Follow these instructions exactly:

1. Read the transcript carefully.
2. Extract only the information that is clearly stated. Do not guess or invent anything.
3. Create the final summary using the exact structure below, but translate the section headers into the language of the transcript.

OUTPUT FORMAT (translate the headers into the transcript language, but keep the structure):

 
A short title describing the meeting.
 
 

<ul>
<li>Summary: </li>
<ul>

<li>Sentence 1</li>
<li>Sentence 2</li>
</ul>
 
 
<li>Decisions:</li>
<ul>
<li>Decision 1</li>
<li>Decision 2</li>

</ul>
 
 
<li>Key Points:</li>
<ul>
<li>Bullet point 1</li>
<li>Bullet point 2</li>
<li>Bullet point 3</li>


</ul>

 
 
<li>Tasks:</li>
<ul>
<li>Person — Task — Deadline (or “No deadline mentioned”)</li>

</ul>
 
 
<li>Open Questions:</li>
<ul>
<li>Question 1</li>
<li>Question 2</li>

</ul>
</ul>

RULES:

Use HTML lists (<ul><li></li></ul>) only for bullet points.
Do not add information that is not in the transcript.
Keep the writing clear and concise.
Keep the structure exactly as shown, but translate the headers.
Do not wrap the whole answer in a single HTML tag.
Use a normal text style. No italic.

TRANSCRIPT:
</pre>

'''3. When meetings are used to take decisions:'''

<pre>
You will receive a meeting transcript. Your task is to turn it into a structured meeting summary.

The entire summary — including section headers, bullet points, and all text — must be written in the same language as the transcript. Detect the language automatically. If the transcript is in german write the summary in german. If the transcript is in english, write the summary in english...

Follow these instructions exactly:

1. Read the transcript carefully.
2. Extract only the information that is clearly stated. Do not guess or invent anything.
3. Create the final summary using the exact structure below, but translate the section headers into the language of the transcript.

OUTPUT FORMAT (translate the headers into the transcript language, but keep the structure):

 
A short title describing the meeting.
 
 

<li>Results:</li>
<ul>
<li>Results 1</li>
<li>Results 2</li>

</ul>
 
 
<li>Miscellaneous:</li>
<ul>
<li>Bullet point 1</li>
<li>Bullet point 2</li>
<li>Bullet point 3</li>


</ul>

RULES:

Use HTML lists (<ul><li></li></ul>) only for bullet points.
Do not add information that is not in the transcript.
Keep the writing clear and concise.
Keep the structure exactly as shown, but translate the headers.
Do not wrap the whole answer in a single HTML tag.

TRANSCRIPT:
</pre>

== Troubleshooting ==
First check if at PBX/Registrations a JSON registration for conference-transcriptions-api is present. If not, check sufficient amount of available PBX-port license - the service requires one.

For troubleshooting the Conference Transcriptions App Service, you need activate the traceflags ''App'', ''Database and'' ''HTTP-Client'' in your App instance.

== Related Articles ==
* [https://sdk.innovaphone.com/16r1/web1/com.innovaphone.conference-transcriptions/com.innovaphone.conference-transcriptions.htm SDK Documentation - Conference Transcriptions API]
* [[{{NAMESPACE}}:Concept App Service myApps Assistant]]
* [[{{NAMESPACE}}:Apps/PbxManager/App myApps ConferenceTranscriptions]]

Reference16r1:Concept App Service Conference Transcriptions

2026-08-14T08:54:33Z

Can:

Howto:DE - Viakom - Voice WB20 TCP SIP-Provider (2026)

2026-08-07T14:07:34Z

Can: Created page with "== Summary == {{Template:SIP_TEST_STATUS_complete|update=August 7th, 2026|url=https://www.viakom.de/produkte/service-providing/professional-service/it-service-support/|productname=Voice_WB20_TCP|providername=Viakom}} === {{SIP_TEST_ISSUES_MR_TITLE}} === {{SIP_TEST_ISSUES_MR_INTRO}} ; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}} ; CLNS : {{SIP_TEST_FACT_CLNS}} ; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}} ; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}} ;..."

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=August 7th, 2026|url=https://www.viakom.de/produkte/service-providing/professional-service/it-service-support/|productname=Voice_WB20_TCP|providername=Viakom}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Viakom-Voice_WB20_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>'Set Calling = Diverting No' recommended in PBX 'Trunk' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Reference:Remote Control Facility

2026-06-26T11:59:15Z

Can: /* Remote Control Facilities and SIP */

===Remote Control Facilities===
==== RCFs implemented in the Phones ====
innovaphone telephones implement certain facilities which sent in during call setup or within a call. These are used for 3rd party phone control, e.g. in myPBX or SOAP (and thus TAPI / CTI applications).

Following remote control facilities are available:
*0 - this default facility is always sent to the calling device if a call is initiated via SOAP, enabling the simulation of an outgoing call on innovaphone device
*1 - reserved
*2 - multicast, sent in setup (UserCall (?) ). Device will accept incoming call right away, will put any active call on hold
*3 - announcement call, sent to the called device, activates hands-free on the phone
*4 - toggle three-party conference state, sent in an active call to the calling device, while a second call on hold exists. Both calls must lie on top of each other on the top of phones display and be available to a conference setup via the menu key on the phone. Any further calls on hold are ignored.
*6 - start call completion (on a busy or disconnected call)
*7 - end call completion (in a new call (?) )
*8 - recording on, requires a recording configuration on the target phone
*9 - recording off
*10 - turn on three-party conference mode (see #4 too)
*11 - turn off three-party conference mode (see #4 too)
*12, 13 - reserved
*14 - mute. Mutes the phone's microphone. Works from phone firmware V9 hotfix 5. Will work only if there is a phone on the call.
*15 - unmute

From phone firmware build 100252, there are some more new codes to control the call mode. They are executed only when the addressed phone is either in handset, headset or handsfree mode,
i.e when calling, connected or disconnected but not when alerting:
*16 - change to handset mode
*17 - change to headset mode
*18 - change to handsfree mode
*19 - monitor mode on (add speaker to handset or headset mode)
*20 - monitor mode off (back to plain handset or headset mode)

From phone firmware build 100567 two more new codes are available. They are executed only when sent in setup (UserCall) instead of the default facility 0, call intrusion is enabled at the phone and the phones state permits to initiate an intrusion.
*21 - intrusion, type silent monitoring
*22 - intrusion, type conference

From phone firmware build 113449 two more new codes are available. They are executed only when the addressed phone is either in handset, headset or handsfree mode and there is a connected call:

*28 - call waiting tone on
*29 - call waiting tone off

From phone firmware build 125059 (v12r2) the following new codes are available:

*30 switch UI to ''home'' app
*31 switch UI to ''phone'' app
*32 switch UI to ''favorites'' app
*33 switch UI to ''call-list'' app
*34 switch UI to ''directory'' app
*35 switch UI to ''config'' app
*36 switch UI to ''video'' app

From phone firmware v15r1

*40 request to perform call hold
*41 request to retrieve held call
*48 recording indication
*49 query to get current audio device (handset,headset,speaker)
*50 answer from phone to pbx with audio device in use

These codes are only valid within an existing call (that is, not as part of a SETUP message).

==== RCFs implemented in the CONF interface ====
Some innovaphone remote control faclities have been implemented in the CONF interface. See [[Reference13r1:Gateway/Interfaces#Innovaphone_remote_controls | ''innovaphone remote controls'' in Reference11r1:Gateway/Interfaces ]].

=== Remote Control Facilities and SIP ===
Remote control facilities are communicatied via H.323 in the PBX. However, the control 1 (connect) can be conveyed via SIP too.
* if the control is sent in an INVITE (outgoing CTI-initiated call), an <code>Answer-Mode: Auto</code> according to RFC5373 is used
* if the control is sent during an alert (accept an incoming alerting call with CTI), a <code>talk</code> event as per RFC3265 is sent to the endpoint:
: <code>NOTIFY ... SIP/2.0</code>
: <code>From: ...</code>
: <code>To: ... </code>
: <code>Call-ID: ...</code>
: <code>CSeq: 434 NOTIFY</code>
: <code>Contact: ...</code>
: <code>Event: talk</code>
This is also supported by innovaphone endpoints (in case of registering them to the PBX via SIP). Note that not all 3rd party SIP devices will support this. Also, some of them support it only if the sender of the INVITE or NOTIFY message authenticates itself to the endpoint. Such devices will not work with the PBX as we do not support PBX authentication towards the endpoint. Some of those devices allow you to disable this restriction though.


Since v15r1sr15/v16r1sr3 Remote control facilities can be communicated also via SIP using SIP INFO message this allow full integration with PhoneAPP/RCCAPP (ex: SwitchboardAPP) of innovaphone.

: <code>INFO sip:endeavour@172.16.71.90:5060;transport=UDP SIP/2.0</code>
: <code>Via: SIP/2.0/UDP 172.16.71.40:5060;branch=z9hG4bK-A24319E0;rport</code>
: <code>From: "Atlantis" <sip:202@test.com;user=phone>;tag=1181559455</code>
: <code>To: <sip:201@test.com;user=phone>;tag=3659098650</code>
: <code>Call-ID: bf8584edf43c6a0178f70090332f68cd@3</code>
: <code>CSeq: 67520993 INFO</code>
: <code>Contact: <sip:202@172.16.71.40:5060;user=phone;transport=UDP></code>
: <code>Content-Length: 12</code>
: <code>Content-Type: innovaphone/remote-control</code>
: <code>Max-Forwards: 70</code>
: <code>...</code>
: <code>control=10</code>

Currently only innovaphone IP Phones can operate with SIP and support the Remote Control facilities. 3rd Party SIP Phones could also support if they implement the remote control facilities in their software.

=== Remote Control Facilities and SOAP ===
You can send remote control facilities on a call with [[Reference12r2:Concept_SOAP_API#UserRc.28integer_call.2C_integer_rc.29 | '''UserRc''' ]] or [[ Reference12r2:Concept_SOAP_API#integer_UserCall.28integer_user.2C_string_cn.2C_string_e164.2C_string_h323.2C_int_reg.2C_InfoArray_info.2C_int_rc.2C_string_srce164.29 | '''UserCall''']].

Reference:Remote Control Facility

2026-06-26T11:58:34Z

Can:

===Remote Control Facilities===
==== RCFs implemented in the Phones ====
innovaphone telephones implement certain facilities which sent in during call setup or within a call. These are used for 3rd party phone control, e.g. in myPBX or SOAP (and thus TAPI / CTI applications).

Following remote control facilities are available:
*0 - this default facility is always sent to the calling device if a call is initiated via SOAP, enabling the simulation of an outgoing call on innovaphone device
*1 - reserved
*2 - multicast, sent in setup (UserCall (?) ). Device will accept incoming call right away, will put any active call on hold
*3 - announcement call, sent to the called device, activates hands-free on the phone
*4 - toggle three-party conference state, sent in an active call to the calling device, while a second call on hold exists. Both calls must lie on top of each other on the top of phones display and be available to a conference setup via the menu key on the phone. Any further calls on hold are ignored.
*6 - start call completion (on a busy or disconnected call)
*7 - end call completion (in a new call (?) )
*8 - recording on, requires a recording configuration on the target phone
*9 - recording off
*10 - turn on three-party conference mode (see #4 too)
*11 - turn off three-party conference mode (see #4 too)
*12, 13 - reserved
*14 - mute. Mutes the phone's microphone. Works from phone firmware V9 hotfix 5. Will work only if there is a phone on the call.
*15 - unmute

From phone firmware build 100252, there are some more new codes to control the call mode. They are executed only when the addressed phone is either in handset, headset or handsfree mode,
i.e when calling, connected or disconnected but not when alerting:
*16 - change to handset mode
*17 - change to headset mode
*18 - change to handsfree mode
*19 - monitor mode on (add speaker to handset or headset mode)
*20 - monitor mode off (back to plain handset or headset mode)

From phone firmware build 100567 two more new codes are available. They are executed only when sent in setup (UserCall) instead of the default facility 0, call intrusion is enabled at the phone and the phones state permits to initiate an intrusion.
*21 - intrusion, type silent monitoring
*22 - intrusion, type conference

From phone firmware build 113449 two more new codes are available. They are executed only when the addressed phone is either in handset, headset or handsfree mode and there is a connected call:

*28 - call waiting tone on
*29 - call waiting tone off

From phone firmware build 125059 (v12r2) the following new codes are available:

*30 switch UI to ''home'' app
*31 switch UI to ''phone'' app
*32 switch UI to ''favorites'' app
*33 switch UI to ''call-list'' app
*34 switch UI to ''directory'' app
*35 switch UI to ''config'' app
*36 switch UI to ''video'' app

From phone firmware v15r1

*40 request to perform call hold
*41 request to retrieve held call
*48 recording indication
*49 query to get current audio device (handset,headset,speaker)
*50 answer from phone to pbx with audio device in use

These codes are only valid within an existing call (that is, not as part of a SETUP message).

==== RCFs implemented in the CONF interface ====
Some innovaphone remote control faclities have been implemented in the CONF interface. See [[Reference13r1:Gateway/Interfaces#Innovaphone_remote_controls | ''innovaphone remote controls'' in Reference11r1:Gateway/Interfaces ]].

=== Remote Control Facilities and SIP ===
Remote control facilities are communicatied via H.323 in the PBX. However, the control 1 (connect) can be conveyed via SIP too.
* if the control is sent in an INVITE (outgoing CTI-initiated call), an <code>Answer-Mode: Auto</code> according to RFC5373 is used
* if the control is sent during an alert (accept an incoming alerting call with CTI), a <code>talk</code> event as per RFC3265 is sent to the endpoint:
: <code>NOTIFY ... SIP/2.0</code>
: <code>From: ...</code>
: <code>To: ... </code>
: <code>Call-ID: ...</code>
: <code>CSeq: 434 NOTIFY</code>
: <code>Contact: ...</code>
: <code>Event: talk</code>
This is also supported by innovaphone endpoints (in case of registering them to the PBX via SIP). Note that not all 3rd party SIP devices will support this. Also, some of them support it only if the sender of the INVITE or NOTIFY message authenticates itself to the endpoint. Such devices will not work with the PBX as we do not support PBX authentication towards the endpoint. Some of those devices allow you to disable this restriction though.


Since v15r1sr15/v16r1sr3 Remote control facilities can be communicated also via SIP using SIP INFO message this allow full integration with PhoneAPP/RCCAPP (ex: SwitchboardAPP) of innovaphone.

: <code>INFO sip:endeavour@172.16.71.90:5060;transport=UDP SIP/2.0</code>
: <code>Via: SIP/2.0/UDP 172.16.71.40:5060;branch=z9hG4bK-A24319E0;rport</code>
: <code>From: "Atlantis" <sip:202@test.com;user=phone>;tag=1181559455</code>
: <code>To: <sip:201@test.com;user=phone>;tag=3659098650</code>
: <code>Call-ID: bf8584edf43c6a0178f70090332f68cd@3</code>
: <code>CSeq: 67520993 INFO</code>
: <code>Contact: <sip:202@172.16.71.40:5060;user=phone;transport=UDP></code>
: <code>Content-Length: 12</code>
: <code>Content-Type: innovaphone/remote-control</code>
: <code>Max-Forwards: 70</code>
: <code>...</code>
: <code>control=10</code>

Currently only innovaphone IP Phones can operate with SIP and support the Remote Control facilities.

=== Remote Control Facilities and SOAP ===
You can send remote control facilities on a call with [[Reference12r2:Concept_SOAP_API#UserRc.28integer_call.2C_integer_rc.29 | '''UserRc''' ]] or [[ Reference12r2:Concept_SOAP_API#integer_UserCall.28integer_user.2C_string_cn.2C_string_e164.2C_string_h323.2C_int_reg.2C_InfoArray_info.2C_int_rc.2C_string_srce164.29 | '''UserCall''']].

Reference:Remote Control Facility

2026-06-26T11:02:33Z

Can: /* RCFs implemented in the Phones */

===Remote Control Facilities===
==== RCFs implemented in the Phones ====
innovaphone telephones implement certain facilities which sent in during call setup or within a call. These are used for 3rd party phone control, e.g. in myPBX or SOAP (and thus TAPI / CTI applications).

Following remote control facilities are available:
*0 - this default facility is always sent to the calling device if a call is initiated via SOAP, enabling the simulation of an outgoing call on innovaphone device
*1 - reserved
*2 - multicast, sent in setup (UserCall (?) ). Device will accept incoming call right away, will put any active call on hold
*3 - announcement call, sent to the called device, activates hands-free on the phone
*4 - toggle three-party conference state, sent in an active call to the calling device, while a second call on hold exists. Both calls must lie on top of each other on the top of phones display and be available to a conference setup via the menu key on the phone. Any further calls on hold are ignored.
*6 - start call completion (on a busy or disconnected call)
*7 - end call completion (in a new call (?) )
*8 - recording on, requires a recording configuration on the target phone
*9 - recording off
*10 - turn on three-party conference mode (see #4 too)
*11 - turn off three-party conference mode (see #4 too)
*12, 13 - reserved
*14 - mute. Mutes the phone's microphone. Works from phone firmware V9 hotfix 5. Will work only if there is a phone on the call.
*15 - unmute

From phone firmware build 100252, there are some more new codes to control the call mode. They are executed only when the addressed phone is either in handset, headset or handsfree mode,
i.e when calling, connected or disconnected but not when alerting:
*16 - change to handset mode
*17 - change to headset mode
*18 - change to handsfree mode
*19 - monitor mode on (add speaker to handset or headset mode)
*20 - monitor mode off (back to plain handset or headset mode)

From phone firmware build 100567 two more new codes are available. They are executed only when sent in setup (UserCall) instead of the default facility 0, call intrusion is enabled at the phone and the phones state permits to initiate an intrusion.
*21 - intrusion, type silent monitoring
*22 - intrusion, type conference

From phone firmware build 113449 two more new codes are available. They are executed only when the addressed phone is either in handset, headset or handsfree mode and there is a connected call:

*28 - call waiting tone on
*29 - call waiting tone off

From phone firmware build 125059 (v12r2) the following new codes are available:

*30 switch UI to ''home'' app
*31 switch UI to ''phone'' app
*32 switch UI to ''favorites'' app
*33 switch UI to ''call-list'' app
*34 switch UI to ''directory'' app
*35 switch UI to ''config'' app
*36 switch UI to ''video'' app

From phone firmware v15r1

*40 request to perform call hold
*41 request to retrieve held call
*48 recording indication
*49 query to get current audio device (handset,headset,speaker)
*50 answer from phone to pbx with audio device in use

These codes are only valid within an existing call (that is, not as part of a SETUP message).

==== RCFs implemented in the CONF interface ====
Some innovaphone remote control faclities have been implemented in the CONF interface. See [[Reference13r1:Gateway/Interfaces#Innovaphone_remote_controls | ''innovaphone remote controls'' in Reference11r1:Gateway/Interfaces ]].

=== Remote Control Facilities and SIP ===
Remote control facilities are communicatied via H.323 in the PBX. However, the control 1 (connect) can be conveyed via SIP too.
* if the control is sent in an INVITE (outgoing CTI-initiated call), an <code>Answer-Mode: Auto</code> according to RFC5373 is used
* if the control is sent during an alert (accept an incoming alerting call with CTI), a <code>talk</code> event as per RFC3265 is sent to the endpoint:
: <code>NOTIFY ... SIP/2.0</code>
: <code>From: ...</code>
: <code>To: ... </code>
: <code>Call-ID: ...</code>
: <code>CSeq: 434 NOTIFY</code>
: <code>Contact: ...</code>
: <code>Event: talk</code>
This is also supported by innovaphone endpoints (in case of registering them to the PBX via SIP). Note that not all 3rd party SIP devices will support this. Also, some of them support it only if the sender of the INVITE or NOTIFY message authenticates itself to the endpoint. Such devices will not work with the PBX as we do not support PBX authentication towards the endpoint. Some of those devices allow you to disable this restriction though.


=== Remote Control Facilities and SOAP ===
You can send remote control facilities on a call with [[Reference12r2:Concept_SOAP_API#UserRc.28integer_call.2C_integer_rc.29 | '''UserRc''' ]] or [[ Reference12r2:Concept_SOAP_API#integer_UserCall.28integer_user.2C_string_cn.2C_string_e164.2C_string_h323.2C_int_reg.2C_InfoArray_info.2C_int_rc.2C_string_srce164.29 | '''UserCall''']].

Reference:Remote Control Facility

2026-06-26T10:27:28Z

Can:

===Remote Control Facilities===
==== RCFs implemented in the Phones ====
innovaphone telephones implement certain facilities which sent in during call setup or within a call. These are used for 3rd party phone control, e.g. in myPBX or SOAP (and thus TAPI / CTI applications).

Following remote control facilities are available:
*0 - this default facility is always sent to the calling device if a call is initiated via SOAP, enabling the simulation of an outgoing call on innovaphone device
*1 - reserved
*2 - multicast, sent in setup (UserCall (?) ). Device will accept incoming call right away, will put any active call on hold
*3 - announcement call, sent to the called device, activates hands-free on the phone
*4 - toggle three-party conference state, sent in an active call to the calling device, while a second call on hold exists. Both calls must lie on top of each other on the top of phones display and be available to a conference setup via the menu key on the phone. Any further calls on hold are ignored.
*6 - start call completion (on a busy or disconnected call)
*7 - end call completion (in a new call (?) )
*8 - recording on, requires a recording configuration on the target phone
*9 - recording off
*10 - turn on three-party conference mode (see #4 too)
*11 - turn off three-party conference mode (see #4 too)
*12, 13 - reserved
*14 - mute. Mutes the phone's microphone. Works from phone firmware V9 hotfix 5. Will work only if there is a phone on the call.
*15 - unmute

From phone firmware build 100252, there are some more new codes to control the call mode. They are executed only when the addressed phone is either in handset, headset or handsfree mode,
i.e when calling, connected or disconnected but not when alerting:
*16 - change to handset mode
*17 - change to headset mode
*18 - change to handsfree mode
*19 - monitor mode on (add speaker to handset or headset mode)
*20 - monitor mode off (back to plain handset or headset mode)

From phone firmware build 100567 two more new codes are available. They are executed only when sent in setup (UserCall) instead of the default facility 0, call intrusion is enabled at the phone and the phones state permits to initiate an intrusion.
*21 - intrusion, type silent monitoring
*22 - intrusion, type conference

From phone firmware build 113449 two more new codes are available. They are executed only when the addressed phone is either in handset, headset or handsfree mode and there is a connected call:

*28 - call waiting tone on
*29 - call waiting tone off

From phone firmware build 125059 (v12r2) the following new codes are available:

*30 switch UI to ''home'' app
*31 switch UI to ''phone'' app
*32 switch UI to ''favorites'' app
*33 switch UI to ''call-list'' app
*34 switch UI to ''directory'' app
*35 switch UI to ''config'' app
*36 switch UI to ''video'' app

From phone firmware v15r1

*40 request to perform call hold
*41 request to retrieve held call
*48 recording indication
*49 query to get current audio device (handset,headset,speaker)
*50 answer from phone to phoneapp/rccapp with audio device.

These codes are only valid within an existing call (that is, not as part of a SETUP message).

==== RCFs implemented in the CONF interface ====
Some innovaphone remote control faclities have been implemented in the CONF interface. See [[Reference13r1:Gateway/Interfaces#Innovaphone_remote_controls | ''innovaphone remote controls'' in Reference11r1:Gateway/Interfaces ]].

=== Remote Control Facilities and SIP ===
Remote control facilities are communicatied via H.323 in the PBX. However, the control 1 (connect) can be conveyed via SIP too.
* if the control is sent in an INVITE (outgoing CTI-initiated call), an <code>Answer-Mode: Auto</code> according to RFC5373 is used
* if the control is sent during an alert (accept an incoming alerting call with CTI), a <code>talk</code> event as per RFC3265 is sent to the endpoint:
: <code>NOTIFY ... SIP/2.0</code>
: <code>From: ...</code>
: <code>To: ... </code>
: <code>Call-ID: ...</code>
: <code>CSeq: 434 NOTIFY</code>
: <code>Contact: ...</code>
: <code>Event: talk</code>
This is also supported by innovaphone endpoints (in case of registering them to the PBX via SIP). Note that not all 3rd party SIP devices will support this. Also, some of them support it only if the sender of the INVITE or NOTIFY message authenticates itself to the endpoint. Such devices will not work with the PBX as we do not support PBX authentication towards the endpoint. Some of those devices allow you to disable this restriction though.


=== Remote Control Facilities and SOAP ===
You can send remote control facilities on a call with [[Reference12r2:Concept_SOAP_API#UserRc.28integer_call.2C_integer_rc.29 | '''UserRc''' ]] or [[ Reference12r2:Concept_SOAP_API#integer_UserCall.28integer_user.2C_string_cn.2C_string_e164.2C_string_h323.2C_int_reg.2C_InfoArray_info.2C_int_rc.2C_string_srce164.29 | '''UserCall''']].

Reference16r1:Concept myApps platform services

2026-06-10T15:49:35Z

Can: /* Codecs */

[[Category:Concept|myApps]]

myApps platform services provide various operating system specific services which can be used by other ''Apps'' running in the [[{{NAMESPACE}}:Concept myApps|myApps client]]. Those services typically are not available in the browser's JavaScript environment and hence must be implemented in native platform code. Therefore, the platform services are installed as native executable on the respective platform.

When myApps is started in a web browser (and hence has no access to the platform services), some Apps will use [https://en.wikipedia.org/wiki/WebRTC WebRTC] services implemented by the browser instead. For ease of reference, features available in this scenario are also described here.

On windows, the platform services also come with their own web browser in which the myApps web App will be started then. This browser is based on google's [https://en.wikipedia.org/wiki/Chromium_(web_browser) Chromium] open source software.
= Applies To =

* [[{{NAMESPACE}}:Concept myApps|myApps]]
* myApps for Windows
* myApps for macOS
* myApps for iOS
* myApps for Android
* myApps Web App (WebRTC)
* myApps for IP270
version 16r1

=Features=
Not all features are available or required on all platforms.
{|
! style="text-align: left; font-weight: bold" | Feature
! style="text-align: left; font-weight: bold" | Description
! style="text-align: left; font-weight: bold"| Availability
|-
| || || Windows || iOS || Android || macOS || Browser<ref>This refers to the myApps web application running in a browser with no platform services available</ref>
|IP270
|-
| [[#Device handling|Audio Devices]]|| manage local audio devices to record and playback audio conversations || ✔ || ✔ || ✔ || ✔ || ✔<ref>audio available but devices managed by web browser</ref>|| ✔
|-
| Video || manage local displays and cameras to capture and render video live stream || ✔ || ✔ || ✔ || ✔ || ✔<ref>video available but devices managed by web browser</ref>|| ✔
|-
| Ringer || manage local ringing device || ✔ || ✔ || ✔ || ✔ || ✔ || ✔
|-
| [[#Application sharing|Application sharing]]
|-
|   presenter || share an application || ✔ || ✗ || ✔ || ✔ || ✔ || ✔
|-
|   consumer || view an application shared by the peer || ✔ || ✔ || ✔ || ✔ || ✔ || ✔
|-
| [[#Hot keys|Hot keys]]|| capture key presses for quick invocation of phone apps (e.g. dial selected number) || ✔ || ✗ || ✗ || ✔ || ✗ || ✗
|-
| [[#URL Handler|tel: and sip: URI handler]]|| intercept clicks on tel: and sip: links in web sites to invoke phone apps || ✔ || ✔ || ✔ || ✔ || ✗ || ✗
|-
| [[#User activity|User activity]]|| set presence state according to user activity || ✔ || ✗ || ✗ || ✔ || ✔<ref>limited, see [[#User activity|User activity]] below</ref> || ✔
|-
| Docking || myApps can be docked persistently to the right or left edge of your screens || ✔ || ✗ || ✗ || ✗ || ✗ || ✗
|-
| Multi-windowing || Apps can be launched in separate windows|| ✔ || ✗ || ✗ || ✔ || ✗ || ✗
|-
| [[#Recording|Recording]]|| Calls can be recorded to recording app|| ✔ || ✔ || ✔ || ✔ || ✗ || ✔
|-
| [[#Notifications|Notifications]]|| ||
|-
|   display notifications || display notifications with OS standard mechanism || ✔ || ✔ || ✔ || ✔ || ✔ || ✗
|-
|   push notifications || receive push notifications while myApps is not running || ✗ || ✔ || ✔ || ✔ || ✔<ref>The browser needs to be running in order to receive push notifications.</ref> || ✗
|-
|   chat and apps || display notifications for chat and other apps || ✔ || ✔ || ✔ || ✔ || ✔ || ✗
|-
|   calls || display notifications for incoming calls || ✔ || ✔ || ✔ || ✔ || ✗<ref>Call notifications are only displayed locally while the phone or softphone app is started.</ref> || ✗
|-
| [[#Local phonebook access|Local phonebook]]|| access local phone book || ✔ || ✔ || ✔ || ✔ || ✗ || ✗
|-
| [[#Microsoft Office Integration|Office presence provider]]|| maps PBX presence state to Microsoft office presence state || ✔ || ✗ || ✗ || ✗ || ✗ || ✗
|-
| [[#Call an external application for calls|External application start]]|| start arbitrary external applications for calls || ✔ || ✗ || ✗ || ✔ || ✗ || ✗
|-
| [[#App Proxy|App Proxy]]|| a caching proxy that provides app persistence || ✔ || ✔ || ✔ || ✔ || ✗ || ✗
|-
| [[#Auto update|Auto update]]|| automatically updates myApps platform services to the same version the PBX has || ✔ || ✔ || ✗ || ✔ || ✔<ref>The then-current web app is always loaded from the PBX upon startup and hence up-to-date by definition</ref> || ✔
|-
| Three party conference || initiate 3-pty-conference using Softphone-App || ✔ || ✔ || ✔ || ✔ || ✗ || ✔
|-
| Exclude VPN || disable use of VPN connections for audio and appsharing || ✔ || ✔ || ✔ || ✔ || ✗ || ✗
|-
| Screen lock || myApps screen lock against unauthorised use || ✗ || ✗ || ✗ || ✗ || ✗ || ✔
|}

<references/>

=Requirements=
* innovaphone PBX 16r1 and up

== Hardware ==
Recommended hardware requirements
* Processor: Dual-core 2Ghz or higher
* RAM: 4 Gb

== myApps for Windows ==
* Windows 11 and up
* Windows Server 2016 and later versions

=== 32 & 64 bit Windows ===
* 32 bit Windows: install the myAppsSetup32.msi from the App Store
* 64 bit Windows: install the myAppsSetup.msi from the App Store
** the 64 bit variant still installs into Program Files (x86), as the main myApps.exe is still a 32bit application
** the 64 bit variant just contains an additional 64 bit binary for the outlook search

=== Windows N editions ===

Windows N editions are missing the ''Media Feature Pack'' which is pre installed on other Windows versions.

Please install the pack from [https://www.microsoft.com/en-us/software-download/mediafeaturepack Microsoft (Windows 10 pack)] before you install myApps. The installer will check if the file <code>C:\Windows\SysWOW64\mfplat.dll</code> exist on your system.

Make sure to install the correct pack depending on your Windows version! There are different packs for Windows 10 1703, 1803, 1809 and 32bit or 64bit etc.

NB: Sometimes the myApps installation will not work even though the media pack is already installed. This is because the installer has no read access to check if the package is already installed. If the above-mentioned file exists and the installer asks to install the Windows Media Feature Pack nevertheless, you have to start the myApps install with administrative rights.

=== Terminal Server environments ===

Audio driver was removed if myApps discovers that it is running in a terminal server environment like Citrix.

The audio driver is needed for the Softphone App but the Softphone App should not use an audio driver at the server side because the audio devices are plugged locally and there would be a delay sending and receiving audio data with the server.

If a customer wants to use the Softphone App at the server side he needs to make use of the myApps Plugin for virtual desktops solution:

[[{{NAMESPACE}}:MyApps_Plugin_for_Virtual_Desktops|Reference15r1:MyApps_Plugin_for_Virtual_Desktops]]

== myApps for macOS ==
* macOS 13 or higher

== myApps for iOS ==
* iOS 16 or higher

== myApps for Android ==
* Android 13 or higher.

== myApps for IP270 ==
Exclusively used in IP270.

= Licenses =
* No license needed for myApps platform services

= Overview =
myApps platform services is a native executable that is installed using the standard mechanisms on the respective operating system. It provides various advanced services which can be used by the myApps web client code as well as the Apps running in the myApps context.

Also, on Windows, the platform services come with their own, dedicated browser to run myApps in. This browser is based on [https://en.wikipedia.org/wiki/Chromium_(web_browser) Chromium]. On iOS, macOS and Android, it is based upon native embedded web view facilities (such as WKWebView) instead.
== Components ==

=== RTP service for audio and appsharing ===
The RTP service provides audio and appsharing as a video stream. VoIP RTP endpoints (e.g. for softphones). It supports STUN, TURN, ICE, SRTP, DTLS. Note however that unlike WebRTC, these endpoints do not ''require'' ICE and DTLS. In other words, they can communicate also with non-compliant (i.e. older) VoIP devices.

Note that the available capabilities when not running the myApps platform services depend on the used browser's WebRTC implementation. See your browser documentation for details.

Apps can request RTP channels using the [https://sdk.innovaphone.com/doc/launcher/Media.htm Media Protocol]'s ''AllocChannel'' message.

==== RTP ports ====
{|
| audio || 50000 -> 50099
|-
| video (app sharing) || 50100 -> 50199
|}

The RTP service will enumerate all local interfaces and create local HOST candidates for ICE. There is an option however to disregard VPN interfaces (more precisely such interfaces with type of ''IF_TYPE_PPP'' or ''IF_TYPE_TUNNEL''). This can eliminate quality issues when RTP data is transmitted through TCP based VPN tunnels.

SRFLX and RELAY candidates are obtained using the STUN and TURN server configuration passed by the App (e.g the ''softphone'' App) as part of the ''AllocChannel'' request.
<code>{"mt":"AllocChannel","channel":"81429cba-396d-43de-8a76-ec020ba8796e","iceServers":[{"urls":"turn:myturn.domaincom:4077?transport=udp","username":"turnuser","credential":"pwd","credentialType":"password"},{"urls":"stun:mystun.domain.com:4077"}],"dn":"Foo Bar","type":"RemoteRtp","kind":"video"}</code>

==== Codecs ====
The installed myApps launchers provide codecs that can be used by softphone apps for media streams. When running in a web browser the codecs depend on the browser version and operating system. See the documentation of your browser for details.

The following codecs are supported:
{|
!style="text-align:left;width:100px;"|Codec
!style="width:100px"|Windows-Launcher
!style="width:100px"|Android
!style="width:100px"|iOS
!style="width:100px"|macOS
!style="width:100px"|Firefox (Browser)
!style="width:100px"|Chrome (Browser)
!style="width:100px"|Edge (Browser)
!style="width:100px"|Safari (Browser)
!style="width:100px"|Opera (Browser)
!style="width:100px"|IP270
|-
|style="text-align:left; background-color:lightgray" colspan="11"|Audio
|
|-
|G711A
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
|G711u
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
|G722
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
|[https://caniuse.com/#search=Opus OPUS-NB]
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
|[https://caniuse.com/#search=Opus OPUS-WB]
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
| colspan="11" style="text-align:left; background-color:lightgray" |Video
|
|-
|[https://caniuse.com/#search=VP8 VP8]
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|-
|-
|[https://caniuse.com/#search=VP9 VP9]
|style="color:green;text-align:center"|✔**
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔**
|style="color:green;text-align:center"|✔**
|style="color:green;text-align:center"|✔**
|style="color:green;text-align:center"|✔**
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔**
|style="color:green;text-align:center"|✔**
|-
|[https://caniuse.com/#search=H264 H264]
|style="color:green;text-align:center"|✔**
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|-
|style="text-align:left; background-color:lightgray" colspan="11"|Application Sharing
|
|-
|Share
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|-
|Watch
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔*
|style="color:green;text-align:center"|✔*
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔
|style="color:red;text-align:center;font-weight:bold;"|X
|style="color:green;text-align:center"|✔
|style="color:green;text-align:center"|✔*
|}

''* small presentation only'' 
''** only for 1:1 calls, not for conferences

==== Video capture ====
The default resolution for video capture is 1920x1080 if available. Otherwise, 1280x720, 640x480, 352x288 or 320x240 will be used. The frame rate is 30 fps if available, otherwise 15 fps. The resulting average bandwidth could reach 1 Mbps.

==== Application sharing ====
Screen content will be transmitted as video stream by the presenter

==== Device handling ====
The RTP service enumerates microphones, loudspeaker, cameras and ringing devices and notifies apps when devices come and go. It is up to the apps using the devices to store preferences.

The RTP service also enables some extended features (such as hook switch or volume control) for supported USB headsets or Bluetooth headsets connected to myApps.
The supported headset-SDKs determine which headset vendors are recommended to be used with the myApps softphone app. 

For this to work, the following vendor specific development kits are integrated in our myApps client. Be aware that the SDK are updated within our Service release :

{| class="wikitable"
|-
! SDK Vendor !! Supported OS !! SDK Version !! innovaphone Service Release
|-
| Jabra|| MacOS || 1.16.4.0 || 14r2sr11
|-
||| Windows || 1.16.4.0 || 15r1sr3
|-
| Epos ''(formerly Sennheiser)'' || MacOS || 12.4.0.5478 || 14r1sr3
|-
||| Windows || n.a. - [[Support:13r3 sr10 MyApps Windows Client - Epos/Sennheiser-Headsets require installed Epos-Connect Software|to be installed separately]]|| 13r3sr10
|-
| Poly ''(formerly Plantronics)'' || MacOS || 3.25.53799.37131 || 13r3sr9
|-
||| Windows || 3.25.53800.37131 || 13r3sr10
|-
| Yealink || MacOS || 3.1.1.23 || 14r2sr1
|-
||| Windows || 3.1.1.23 || 14r2sr2
|}

Notes:
* It is possible to inhibit the start of the Sennheiser SDK (SenncomSDK.exe) using the <code>DISABLEHEADSETS</code> directive of the installer (see [[#MSI Parameters and install options| MSI parameters]] below).

* Starting with V13r3sr10, the Epos-SDK needs to be installed separately using the Epos Connect software to ensure full compatibility between current Epos headset models and native myApps-Windows client. For details [[Support:13r3 sr10 MyApps Windows Client - Epos/Sennheiser-Headsets require installed Epos-Connect Software|refer to this article]].


myApps-IP270 supports use of [[Reference9:Concept USB Headset|USB devices known for innovpahone desk phones]].

==== Ring tones ====
Ring tones can be played. Apps can choose the tone from a pre-defined list of ring tones.

On Windows, custom ring tones can be uploaded as .mp3 files to the <code>ringtones</code> sub-directory of myApps' roaming directory (which usually is in <code>%appdata%\innovaphone\myApps\ringtones</code>).

On Android, custom ring tones can be added to the system via Android settings.

On iOS, custom ring tones can be uploaded as .mp3 files to the <code>Ringtones</code> subdirectory of the myApps file share that is available in iTunes if the iPhone has been connected via USB.

On macOS, custom ring tones can be uploaded as .mp3 files to <code>~/Library/Containers/com.innovaphone.client-macos/Data/Documents/Ringtones</code>.

==== Debugging ====
For extended debugging, turn on the ''Audio'', ''Media'' and ''AppSharing'' traces in myApps.
=== Hot keys ===
On Windows and macOS systems, myApps platform services can listen for hot keys and invoke certain functions. Invocation is done by sending API messages to myApps which passes it to an appropriate API provider (in the cases described here, this will be a ''phone'' or ''softphone'' or ''rcc'' App typically. See [[{{NAMESPACE}}:Concept_myApps#Client_APIs_and_default_apps | Client APIs and default apps]] for more details about this mechanism.

The hot keys can be specified using the ''advanced settings'' user interface (see [[#UI elements| UI elements]] below. Any of the function keys F1 to F11 (optionally combined with up to two modifier keys ''alt'', ''ctrl'', ''shift'' or ''win'') can be chosen for each function. If you do not want to start the call with "Hotkey+Enter" because you would have to wait for the focus, the hotkey can also be pressed twice and the number is dialled directly.

; dial selected number : Initiates a call using the currently selected text as target.

: A ''PrepareCall'' message with the ''text'' argument set to the selected text and the ''adjust'' argument set to <code>true</code> will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm ''com.innovaphone.phone'' API].

:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.phone","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":"mt":"PrepareCall","text":"13","adjust":true}}</code>

; accept call : Accepts a currently alerting call.

: A ''ConnectCall'' message will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm ''com.innovaphone.phone'' API].

:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.phone","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":{"mt":"ConnectCall"}}</code>

; reject/disconnect call : Rejects a currently alerting call or disconnects an active call.

: A ''DisconnectCall'' message will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm ''com.innovaphone.phone'' API].

:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.phone","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":{"mt":"DisconnectCall"}}</code>

=== URL Handler ===

On Windows systems, two URI-handler are installed with the myApps platform services. Windows will call up this URI handler when a user clicks on an appropriate link, for example in a web site.

The handler will the send an API message to myApps which passes it to an appropriate API provider (in the cases described here, this will be a ''phone'' or ''softphone'' or ''rcc'' App typically. See [[{{NAMESPACE}}:Concept_myApps#Client_APIs_and_default_apps | Client APIs and default apps]] for more details about this mechanism.

; tel URI : call a number, e.g. <code>tel:4711</code>

: A ''PrepareCall'' message with the ''num'' argument set to the selected text and the ''adjust'' argument set to <code>true</code> will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm ''com.innovaphone.phone'' API].
:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.phone","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":{"mt":"StartCall","num":"4711","adjust":true}}</code>
; sip URI : call a SIP name, e.g. <code>sip:zkl@innovaphone.com</code>

: A ''PrepareCall'' message with the ''sip'' argument set to the selected text and the ''adjust'' argument set to <code>true</code> will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm ''com.innovaphone.phone'' API].
:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.phone","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":{"mt":"StartCall","sip":"zkl@innovaphone.com","adjust":true}}</code>
; im URI : start chat with SIP name, e.g. <code>im:zkl@innovaphone.com</code>

: A ''StartChat'' message with the ''sip'' argument set to the selected text will be sent to the [http://sdk.innovaphone.com/web1/com.innovaphone.chat/com.innovaphone.chat.htm ''com.innovaphone.chat'' API].
:: <code>{"mt":"ApiRequest","apiId":"com.innovaphone.chat","consumer":"@local-ae2fc2ab74-3f1e-4ab9-b215-d42f213520317","msg":{"mt":"StartChat","sip":"zkl@innovaphone.com"}}</code>

On macOS systems myApps might be made the default application to handle tel URI e.g. <code>tel:4711</code> via Apple FaceTime. Open the "FaceTime" menu "Settings..." and select myApps as "Default for phone calls".

On iOS ''tel'' URIs are always dialed via GSM. Therefore myApps iOS also reacts to URI schemes ''com.innovaphone.tel'', ''com.innovaphone.sip'' and ''com.innovaphone.im'', e.g. <code><nowiki>com.innovaphone.tel:4711</nowiki></code>, <code><nowiki>com.innovaphone.sip:zkl@innovaphone.com</nowiki></code>, <code><nowiki>com.innovaphone.im:zkl@innovaphone.com</nowiki></code>.

=== User activity ===
On Windows and macOS systems, the myApps platform services can monitor user keyboard/mouse activity and change the user's presence state after a certain amount of inactivity. The timeout can be specified using the ''advanced settings'' user interface (see [[#UI elements| UI elements]] below.

myApps will then send a [https://sdk.innovaphone.com/doc/appwebsocket/myApps.htm#SetUserActivity''SetUserActivity''] message to the PBX using the ''myApps'' protocol.

: <code>{"mt":"SetUserActivity","inactive":true}</code>

This will change the ''status'' property of the ''im:'' contact for the user's own presence and hence result in a presence update from the PBX to myApps

: <code>{"mt":"UpdateOwnPresence","presence":[{...},{"contact":"im:","activity":"","status":"closed"}]}</code>
The ''closed'' status is reflected in the grey status color when displaying a contact [[Image:myapps-inactive.png|myapps-inactive.png/|myapps-inactive.png/]].

On iOS and Android, the state is set to ''inactive'' as soon as the App is brought to background.
When myApps platform services are not available (i.e. when running the web application in a browser solely) a limited user activity monitoring is available: the state is set to active when the web page is not used for more than 5 minutes.

=== Recording ===

The new launcher offers the possibility to record the audio of incoming and outgoing calls. In order to activate that functionality the URL of the recording instance must be configured in either the PBX (PBX->myApps->Config: Recording URL) or the softphone App (Settings->Audio Recording (URL))

[[Image:PBX-Recording-Settings.png|pbx-recording-settings.png/|pbx-recording-settings.png/]] [[Image:Recording-Softphone-Settings.png|recording-softphone-settings.png/|recording-softphone-settings.png/]].

As long as that URL is configured the audio data of all calls are stored as pcap-files under that URL.
If the URL points to a CF device in the PBX, write access must be granted for that URL (PBX->Services->HTTP->Server:Public compact flash access) and if the URL points to the recording app, the files can be accessed via the recording app [[{{NAMESPACE}}:Concept_App_Service_Recordings|recording]].

Under PBX->myApps the administrator can set a certain default behaviour of the audio recording like whether or not the recording should start automatically at the beginning of the call (Recording by Default ON/OFF), only calls with external numbers should be recorded (Record external calls only) or whether or not the user should be able to start/stop the recording himself (Allow user incall recording control). Except for the last parameter these parameters can also be modified by the user in its softphone settings if the administrator doesn't set the FORCE flag.

If the user was allowed by the admin to control the recording a recording switch is active during the call when the "Media" Panel is opened. There the audio recording may be stopped and continued at will. A red recording notice is shown in the top right corner when the recording actually takes place.

[[Image:Recording-incall-switch.png|recording-incall-switch.png/|recording-incall-switch.png/]]

=== Notifications ===

The myApps platform services can use the OS specific notification mechanism (e.g. ''desktop notifications'' on Windows) to display messages (e.g. ''incoming new chat message'') to the user.

Note that the actual rendering of the notification is under control of the OS. Therefore, myApps must be allowed to show notifications and its appearance can be restricted by OS native settings.

==== Microsoft Windows Notifications ====

Microsoft Windows Server editions (2016, 2019, 2022) are just capable of showing a single ''IncomingCall'' notification at the same time (we couldn't find a workaround for this limitation). 
An ''IncomingCall'' notification is visible the whole time instead of being moved to the action center after a certain time. 
 
A notification about a missed call uses the ''IncomingCall'' type so that this notification is visible until the user returns. 
Due to the above limitation, on a new arriving call such a missed call notification is transformed to a default notification which will be moved to the action center automatically. 
 
On non server editions, you can have multiple IncomingCall notifications at the same time (so two parallel incoming calls will be indeed notified at the same time), but the missed call notification handling is the same on both platforms!

Thus there will be always just '''one''' missed call notification visible and previous missed calls can be found inside your action center!

To see myApps notifications, ensure:
* System -> Notifications
** enable notifications
** disable "Do not disturb" or allow myApps as priority application while "Do not disturb" is active
** enable notifications for myApps in the list of applications
* System -> Focus
** if a focus session is active and the "Do not disturb" is activated during a focus session, make sure that myApps is a priority application (see above)

==== macOS Notifications ====
Notifications are the same as on Windows.
The difference is, that for macOS, notifications need to be allowed in the system settings.
Go to Notifications - myApps, select Banner and enable all check marks.

=== Local phonebook access ===
'''Contact Search:''' The myApps platform services implement an ''API provider'' for the [http://sdk.innovaphone.com/web1/com.innovaphone.search/lib1_api_search.htm ''com.innovaphone.search'' API]]. They perform search capabilities on the OS' local phone books which can be used by Apps like the ''phoneapp''.

Apps would send a ''Search'' request to the API:

: <code>{"mt":"ApiRequest","consumer":"dev:SwPh_zkl_5e42e884","provider":"*","src":"4","msg":{"mt":"Search","type":"contact","search":"john doe"},"apiId":"com.innovaphone.search"}</code>
Search results are delivered as ''SearchInfo'' messages:

: <code>{"mt":"ApiResult","src":"3","provider":"@local-8125d22e37-519d-4056-bfe5-c52ef2ae8fabb0","consumer":"dev:SwPh_zkl_5e42e884","client":"@client-f62702dd86-be3f-47fc-b4bc-7a21627b75b2ea","msg":{"mt":"SearchInfo","relevance":2000,"adjust":true,"type":"contact","contact":{"givenname":"John","sn":"Doe","company":"ACME","position":"Head of everything","telephonenumber":["11111","22222"],"homephone":["+4944444","33333"],"mobile":["+49 (123) 55555"]}},"api":"com.innovaphone.search"}</code>

'''Reverse Lookup:''' The myApps platform services implement an ''API provider'' for the [http://sdk.innovaphone.com/web1/com.innovaphone.phonelookup/lib1_api_phonelookup.htm ''com.innovaphone.phonelookup'' API]. They perform search capabilities on the OS' local phone books which can be used by Apps like the ''phoneapp''.

Apps would send a ''Lookup'' request to the API:

: <code>{"mt":"ApiRequest","consumer":"dev:SwPh_zkl_5e42e884","provider":"*","src":"4","msg":{ mt: "Lookup", prefixIntl: "000", prefixNtl: "00", prefixExt:"0", area: "7031", country: "49", lookup: "0004970311234567" },"apiId":"com.innovaphone.lookup"}</code>

Search results are delivered as ''LookupInfo'' messages:

: <code>{"mt":"ApiResult","src":"3","provider":"@local-8125d22e37-519d-4056-bfe5-c52ef2ae8fabb0","consumer":"dev:SwPh_zkl_5e42e884","client":"@client-f62702dd86-be3f-47fc-b4bc-7a21627b75b2ea","msg":{mt: "LookupInfo", dn: "Jake Blues", contact: { telephonenumber: ["0004970311234567"], givenname: "Jake", sn: "Blues", company: "Blues Brothers" </code>

==== Windows ====
On Windows, the search and lookup are performed in all of the user's Outlook contact folders. As opposed to the search implemented in the ''Contacts'' and ''Users'' App, all items are returned which match any of the search words (i.e. searching for ''a b'' will return items matching either ''a'' or ''b'').

; searched properties : firstname, lastname
; returned properties : Following Outlook contact phone number properties are returned (if available):

:* OFFICE_TELEPHONE_NUMBER as ''telephonenumber''
:* OFFICE2_TELEPHONE_NUMBER as ''telephonenumber''
:* HOME_TELEPHONE_NUMBER as ''homephone''
:* HOME2_TELEPHONE_NUMBER as ''homephone''
:* MOBILE_TELEPHONE_NUMBER as ''mobile''
:* BUSINESS_FAX_NUMBER as ''facsimiletelephonenumber''
Note that contact information is cached in the search provider. Updated contacts may therefore become effective after a while only.
Outlook search will create its own trace file <code>myAppsOutlookSearch-</code>''date-time''<code>.txt</code> in the standard trace directory.

This search provider is always installed and can be disabled. There is no need (nor possibility) to enable it in the ''Apps'' tab of the PBX's user object. Also, no ''App'' object needs to be created for it.

==== Android/iOS ====
The search and lookup are performed in the contacts.

==== macOS ====
The search and lookup are performed in the contacts. If you wish to disable local contact lookup, go to system settings - Security & Privacy and disable the access to contacts for myapps.
=== Microsoft Office integration ===

The myApps platform services has a ''office presence provider'' that can provide the user's presence state to Office applications. See [[{{NAMESPACE}}:Concept_myApps_Office_Integration|myApps Office Integration]] for details.

This feature is installed by default. However, it can be disabled using the ''OFFICEPRESENCE'' MSI Parameter. Also, a check-mark is available in the setup dialog.

=== Call an external application for calls ===

Phone Apps (such as the phoneapp or softphone) can initiate the start of an external application when a new call appears (either incoming or outgoing). The actual spawning of the application is done by the myApps platform service. Also, the application properties (such as e.g. the executable's path) is configured in the myApps platform services (see [[#UI elements|Advanced settings]] in the ''UI elements'' section below).

A number of arguments can be passed to the application by substituting $-variables in the ''Parameter'' field:

; $n : phone number as dialed (called party number for outgoing calls) or received (calling party number for incoming calls)

; $N : called or calling party number in ''national'' format (e.g. 07031730090)

; $I : called or calling party number in ''international'' format (e.g. +497031730090)

: note that both $N and $I only work if $n includes both subscriber number and area code (e.g. 07031730090). Otherwise they are equal to $n

; $d : display name of peer (if known)

; $u : URI name of the peer (if available eg with a federation call)

; $c : conference id

: this is a globally unique ID for this call and may be used to relate the call to the ''guid'' found in the CallInfo structure in the [http://wiki.innovaphone.com/index.php?title=Reference10:SOAP_API#CallInfo SOAP-API] and [http://sdk.innovaphone.com/doc/appwebsocket/RCC.htm RCC-API ]. Also, corresponding [[Reference10:Call Detail Record CDR PBX|CDRs]] can be related using the ''event'' tag's ''conf'' attribute.
The start of an external application can be requested using the ''com.innovaphone.externalapps'' API.

Some setup examples are [[Howto:Integrate External Apps in innovaphone UC clients|shown here]].

=== Push ===

Mobile operating systems usually inhibit network operation of apps which run in the background or are closed by the user. This is done in order to reduce battery consumption. Unfortunately, this also stops such apps to maintain a registration by regularly sending ''keep alive'' messages to a server (in our case to the PBX). As a result, myApps will be disconnected from the PBX. When the PBX determines that there is an event for the application which needs a response, it needs to wake up the app using a dedicated channel provided by the operating system. This mechanism is know as ''push''. When running on iOS or Android, myApps supports ''push''.

For ''push'' to work, a [[{{NAMESPACE}}:PBX/Objects/Push|''push object'']] needs to be configured in the PBX . Also, it needs to be enabled on the mobile phone for the myApps app.
This mechanism is quite similar in v12 and v13, so you can refer to [[{{NAMESPACE}}:Concept_Push_Notifications_for_iOS_and_Android|the concept for push notifications for iOS and Android]] for more details.

Also, helpful hints can be found in [[Howto:Troubleshoot v13 Push with myApps for Android and iOS]].

=== App Proxy ===

myApps runs further ''Apps'' (such as e.g. the ''phoneapp'') as a web page in an IFRAME of the browser myApps is running in. The App's page code is loaded either from the PBX or from an ''application platform'' (AP). This however would mean that the App's IFRAME would remain empty (a dead white screen) when the PBX or AP is not available. To make sure the App can start-up anyway, the myApps platform services feature the so-called ''App Proxy''. This is a caching proxy that caches all the App code so it is available even in case of network failure. When myApps runs in the context of the platform services, Apps are therefore not loaded from the App source directly, but from the local App proxy.

The cached files are stored in the PCs local file system in the <code>%LOCALAPPDATA%\innovaphone\myApps\appproxy</path></code>. There is no configuration required. However, if myApps seems to run with outdated or corrupt cached copies of the App, you can safely delete the entire directory.

=== Auto update ===

On Windows and on macOS, the myApps platform services can auto-update themselves to a common version. This is controlled by the [[{{NAMESPACE}}:PBX/Config/myApps#Launcher_Software_Update | ''Launcher Software Update'']] settings under ''PBX/Config/myApps'' in the PBX.

When myApps is started or the user logs in or myApps needs to re-connect to the PBX, the platform services will use the [http://sdk.innovaphone.com/web1/com.innovaphone.client/lib1_api_client.htm com.innovaphone.client API] to learn the desired version (''launcherUpdateBuild'', which is part of the API's ''model''). If this differs from the current version, the platform services will try to download the respective new version.

{
"mt": "ApiUpdate",
"apis": {
"com.innovaphone.client": {
"@client": {
"title": "innovaphone myApps",
"model": {
"launcher": true,
"launcherUpdateBuild": "134906",
"appStoreUrl": "http://store.innovaphone.com/release/download/"
}
}
}
}
}

The installation of the downloaded version is done by the ''innovaphonemyAppsUpdateService''. This service is installed and enabled during the initial installation of the myApps platform services. To disable auto-update, either leave the ''Launcher Software Update'' settings empty or set the service's start mode to ''disabled'' in the Windows ''services control panel''.

Note that on Windows the update service does not work on terminal servers. Administrators must do myApps base services updates using standard windows mechanisms.

Note that on macOS if myApps has been installed from the Apple Store it is assumed that auto update from the PBX is not desired and disabled therefore. If you need to activate the auto update later, you will need to reinstall myApps using the .pkg file.
On Android/iOS/macOS updates can be downloaded from the respective app store.

The ''Devices'' app can not update software installed on Windows PCs directly. However, when the PBX is updated using an ''update job'' in the ''Devices'' App, the ''Launcher Software Update'' settings will be updated accordingly and hence the myApps base services will ultimately also be updated to the same version.

==== Auto update flow on Windows ====

* On start of myApps, myApps checks if an update is available and ready for installation
** if yes, the update is installed directly, without user interaction (a popup is shown during the installation)
** if not, myApps starts
* if an update is available while myApps is already running, an update notification will be shown which let's the user choose to install the update now or later (the notification will then popup again after one hour)

==UI elements ==
There are a few user interfaces provided by the platform services:
===tray-icon (Windows only) ===
::[[Image:myapps-tray.png|myapps-tray.png/|myapps-tray.png/]]
:Allows to
:* terminate myApps
:* toggle the ''autostart'' state
:* toggle the ''show in task bar'' state
:* open the trace folder
:
=== PBX connect form===
:: [[Image:myapps-connect.png|myapps-connect.png/|myapps-connect.png/]]
: Allows the user to specify the connect data for the PBX (i.e. IP address or DNS name)
:
=== Advanced settings===
[[Image:myapps-settings0.png| More button to access advanced settings]]
<gallery mode="packed" heights="420" style="float:left">
Image:MyApps-settingsNew.png|Advanced generic configuration
Image:MyApps-settingsNew2.png|Hotkeys configuration
Image:MyApps-settingsNew3.png|External launcher configuration
Image:MyApps-settingsNew4.png|Trace configuration
</gallery>
<div style="clear:both;"></div>

: Allows modification of various platform dependent settings (such as the hotkey selection on Windows, external application actions, etc).

== Interfaces ==
=== Provided APIs ===

; [http://sdk.innovaphone.com/web1/com.innovaphone.search/lib1_api_search.htm com.innovaphone.search] : access to local phone book entries by the [[#Local phonebook access|Local phonebook access]] component.
; [http://sdk.innovaphone.com/web1/com.innovaphone.launcher/com.innovaphone.launcher.htm com.innovaphone.launcher] : display of OS specific user notifications and receipt of related user actions
; com.innovaphone.notificationhandler : reports back click on a notification.
; com.innovaphone.externalapps : to start external applications, see [[#Call an external application for calls|Call an external application for calls]] above

=== Used APIs ===

; [http://sdk.innovaphone.com/web1/com.innovaphone.phone/com.innovaphone.phone.htm com.innovaphone.phone] : used to initiate new or manipulate existing calls by the [[#Hot keys|Hot keys]] and [[#URL handler|URL handler]] components.

; [http://sdk.innovaphone.com/web1/com.innovaphone.chat/com.innovaphone.chat.htm com.innovaphone.chat] : used to start a new chat by the [[#URL handler|URL handler]] component.

; [http://sdk.innovaphone.com/web1/com.innovaphone.client/lib1_api_client.htm com.innovaphone.client] : the model is used to learn the update settings, see [[#Auto update|Auto update]] above

=== Protocols ===

; [https://sdk.innovaphone.com/doc/launcher/Media.htm Media Protocol] : used by apps to allocate RTP channels, see [[#RTP service for audio.2C video and data|RTP service for audio, video and data]] above

== Related App Services ==

none

== Known limitations ==
; Incoming call as banner on myApps for iOS : Since iOS 14 the iOS CallKit presents incoming calls as a banner leaving the original green answer button of myApps visible. Use only the blue button of the banner to accept the call or change iPhone Settings, App "Phone", "Incoming Calls" to "Full Screen" to hide the myApps user interface again during call answering.

; Call answer in speakerphone mode even with active Bluetooth headset on myApps for iOS : This causes unwanted speakerphone operation if the smartphone is used with a Bluetooth car audio system. The behaviour can be changed by selecting ''Bluetooth Headset'' in this setting:
:''iOS Settings->Accessibility->Touch->Call Audio Routing: Automatic / Bluetooth Headset / Speaker''
:''iOS Einstellungen->Bedienungshilfen->Tippen->Anrufaudioausgabe: Automatisch / Bluetooth-Headset / Lautsprecher''

; Windows Server 2016 (Windows 10 Build 1607) : windows just shows the first notification. Further notifications aren't displayed until the previous ones are removed from the notification center. Current windows builds do not show this behaviour anymore.

; Problems on Mac computers with Yealink USB headsets
: we have received reports that myApps quits unexpectedly on some Mac computers when a Yealink headset is plugged in. Unfortunately, we could not find out the cause yet. If you use Yealink USB headsets and have a similar issue, please open a support ticket and send myApps traces.

; Poly / Plantronics headset buttons only functional if myApps is started with Rosetta
: myApps macOS supports Apple M1/M2 hardware natively. However, the Poly / Plantronics headset SDK is only available for Intel platform and thus myApps needs to be started via Apple's Intel emulator Rosetta if a Poly / Plantronics headset is used. This is done with right-click on the myApps executable, ''Information'', ''Open with Rosetta''.

; Windows surface devices may not work correctly
: Chromium does not get touch keyboard events. USB Keyboards may not be recognized either.

= Installation =

== Windows ==

myApps platform services are installed on Windows using the .msi file found in the ''myApps Windows'' package from [https://store.innovaphone.com/release/download.htm store.innovaphone.com].

myApps can update itself automatically, see [[#Auto update|Auto update]] above.

=== MSI Parameters and install options ===

The MSI installer of myApps for Windows supports the following parameters and can be edited with [https://docs.microsoft.com/en-us/windows/win32/msi/orca-exe Microsoft Orca]. You can add your parameters in the table ''property''.

; SERVER (REG_SZ): the PBX's server address (without protocol like https://)
; OFFICEPRESENCE (REG_DWORD): '''false''' to disable presence integration in Microsoft Office
: this is also available as a check-mark when running the install manually

; DISABLEHEADSETS (REG_DWORD): '''true''' to disable headsets support, see [[#Device handling|Device handling]] above

; EXTERNALAPPS (REG_SZ): pre-define external applications, see [[#Call an external application for calls|Call an external application for calls]] above
: e.g. <code>"{""externalApps"":[{""id"":0,""name"":""Wireshark"",""path"":""C:\\Program Files\\Wireshark\\Wireshark.exe"",""param"":""test $I""}]}"</code>

; FORCERESTART (REG_DWORD): '''true''' (or any string ...) kills myApps during the installation and restarts it for the currently logged in user, if it was running

; DISABLELOCALHOST (REG_DWORD): '''true''' to disable use of '''localhost''' string to access the local webserver. Use '''127.0.0.1''' instead

; EXCLUDEINTERFACES (REG_SZ): some VPN interfaces are not detected by Windows as IF_TYPE_PPP or IF_TYPE_TUNNEL and therefore the '''media outside VPN''' setting is not taking effect. With this option interfaces can be pre-defined that will not be used for media. Interfaces must be comma separated
: e.g. <code>EXCLUDEINTERFACES="172,192.168,10.10"</code>

; FASTERDOWNLOADS (REG_DWORD): '''true''' to have faster downloads without artificially slowing down the download of an update (which is done to avoid audio lags if clients have slow networks).

Current settings are stored in the registry at <code>Computer\HKEY_CURRENT_USER\Software\innovaphone\myApps</code> or at <code>Computer\HKEY_LOCAL_MACHINE\SOFTWARE\WOW6432Node\innovaphone\myApps</code>

Boolean values like OfficePresence are stored in registry entries with type REG_DWORD and values 1 or 0. 0 disables the setting and 1 enables it.

== iOS ==

myApps platform services are installed on iOS by loading ''innovaphone myApps'' from the ''App Store''.

=== Configuration via MDM ===

the PBX's server URL can be pre-configured by specifying this dictionary in the MDM
<plist>
<dict>
<key>server</key>
<string>pbx.example.com</string>
</dict>
</plist>

== macOS ==

myApps platform services might be installed directly from the Apple store. An installer package <code>myapps.pkg</code> and a disk image <code>myapps.dmg</code> is also available from the innovaphone app store. Install <code>myapps.pkg</code> by double-click on the file and follow the instructions of the installer. myApps becomes available in the Applications folder and can be opened by double-click. Or download and open <code>myapps.dmg</code> and double klick myApps. If desired integrate it into the app dock by right click, ''Options, Keep in the dock''.

If installed from the innovaphone app store, myApps can update itself automatically, see [[#Auto update|Auto update]] above.

If installed from the Apple store, macOS notifies about updates on the Apple store. myApps [[#Auto update|Auto update]] is disabled then.

If a clean-install of the client is necessary, the folder "/Users/username/Library/Containers/myapps" needs to be deleted. To be on the safe side also delete it from the trash bin.

=== Configuration via MDM ===

the PBX's server URL can be pre-configured by specifying this dictionary in the MDM
<plist>
<dict>
<key>server</key>
<string>pbx.example.com</string>
</dict>
</plist>

=== Preferences ===

macOS supports preference settings that can be set via a shell command or via Mac remote management

defaults write com.innovaphone.client-ios-14r1 server "PBX-server-URL"

The following parameters and can be set through this method:

; server: the PBX's server URL

=== Setting myApps as Default App for SIP-URLs ===

{| class="wikitable"
|defaults write com.apple.LaunchServices/com.apple.launchservices.secure LSHandlers -array-add '{
LSHandlerURLScheme = sip;
LSHandlerRoleAll = "<CFBundleIdentifier>";
}'
|}

To find the “CFBundleIdentifier”, proceed as follows:
* In the Finder under “Applications”, search for the desired myApps client that you want to set as the default app.
* Right-click on “Show package contents” -> you will find the “CFBundleIdentifier” in the Info.plist file.

A restart of the MAC is required.

=== Using Sennheiser headsets ===
If you use Sennheiser headsets, you should also install the then-current <code>DSEA_SDK_v</code>''version''<code>.pkg</code> package, after you installed the myApps client. Without that, audio will still work, but not the controls on the headset. You will need to keep that up-to-date yourself, as it is not updated by myApps's auto-update function.

== Android ==

myApps platform services are installed on Android by loading ''innovaphone myApps'' from the ''Play Store''.

=== Configuration via MDM ===

the PBX's server URL can be pre-configured by specifying a property "server" with string value "pbx.example.com" in the MDM.

== IP270 ==
For configuration instructions, refer to the [[Reference16r1:Concept IP270|IP270 concept article]].

= Configuration =

== Server configuration ==
When opening myApps for the first time, the user is prompted for the Server. Usually only the hostname (DNS host name or IP address) needs to be configured.

But there are more options for special PBX configurations.

; Non-standard HTTPS port
: If the PBX uses a non-standard HTTPS port, it must be appended to the host name separated by a colon (<code>:</code>).
: Example: <code>pbx.example.com:4444</code> (expands to <code>https://pbx.example.com:4444/PBX0/APPCLIENT/appclient.htm</code>)
; DynPBX module name
: If the PBX is a DynPBX, the module id must be appended to PBX0 separated - (<code>/</code>).
: Example: <code>pbx.example.com/PBX0-1</code> (expands to <code>https://pbx.example.com/PBX0-1/APPCLIENT/appclient.htm</code>)
; Softphone physical location
: If user defined physical location shall be used for softphone, you can append it using a parameter <code>#phys=</code>.
: Example: <code>pbx.example.com#phys=slave</code> (expands to <code>https://pbx.example.com/PBX0/APPCLIENT/appclient.htm#phys=slave</code>)

Example 1: PBX pbx.example.com with standard configuration
pbx.example.com

Example 2: PBX slave.example with DynPBX module ID 1, HTTPS port 4444 and physical location master
slave.example.com:4444/PBX0-1#phys=master

=== HTTP proxy support ===

myApps platform services do support operation via HTTP proxy now. If one or more proxies have been configured in the network settings of the operating system for the active network connection, HTTP CONNECT tunnels are established.

On Windows user name and password can be specified for the tunnel servers as generic credentials in the credentials manager (Anmeldeinformationsverwaltung). The name of the credentials must be the tunnel server hostname.

On Android user name and password can be specified through Android ''Settings, Accounts'' by adding a myApps ''HTTP Proxy Credentials'' account. The name of the account must be the tunnel server hostname.

== Platform specific settings ==
When myApps runs under the myApps platform services, it will show various platform specific settings as part of its ''burger menu'', so the user can set them. See ''Advanced settings'' in [[#UI elements|UI elements]] above.

Some options can also be set globally for all myApps clients in the PBX's [[{{NAMESPACE}}:PBX/Config/myApps#Client_Settings|PBX/Config/myApps ''Client Settings'']]
{|
! style="text-align: left; font-weight: bold" | Option

! style="text-align: left; font-weight: bold" | Description

! style="text-align: left; font-weight: bold" | Where to set

!
! style="text-align: left; font-weight: bold"| Availability
|-

| || || User menu || PBX ''Client Settings'' || Windows || iOS || Android || macOS
|IP270<ref>myApps-IP270 offers device specific settings explained in [[Reference16r1:Concept IP270|IP270 concept article]]</ref>
|-
| Autostart || Launch myApps on login || ✔ ||✔ ||✔ || ✗ || ✗ || ✔
|✗
|-

| Appear offline after || controls after which idle time a user is considered ''inactive''. See [[#User activity|User activity]] above || ✔ ||✔ ||✔ || ✗ || ✗ || ✔
|✗
|-

| Hotkeys || Hotkeys for call dial, accept, reject. See [[#Hot keys|Hot keys]] above || ✔ ||✔ ||✔ || ✗ || ✗ || ✔
|✗
|-

| Docking || Docking mode (left, right, none). See [[#???|??]] above || ✔ ||✔ ||✔ || ✗ || ✗ || ✗
|✗
|-

| Desktop notifications|| Turn on/off platform notifications. See [[#Notifications| Notifications]] above || ✔ ||✔ ||✔ || ✗ || ✗ || ✔
|✗
|-

| VPN || Disable VPN address for ICE candidate selection. See [[#RTP ports| RTP ports]] above || ✔ ||✔ ||✔ || ✗ || ✔ || ✔
|✗
|-

| Show in taskbar|| Show myApps in the taskbar in addition to it's tray icon. || ✔ ||✗ ||✔ || ✗ || ✗ || ✗
|✗
|-

| Log flags || turn on/off certain trace levels. See [[#Troubleshooting|Troubleshooting]] below. || ✔ ||✔ ||✔ || ✔ || ✔ || ✔
|✔

|-

| External applications || define the applications available for Apps to be started. See [[#Call an external application for calls|Call an external application for calls]] above. || ✔ ||✗ ||✔ || ✗ || ✗ || ✔
|✗
|-

| Ring in headset || send ring tone for incoming to headset instead of loudspeaker. || ✔ ||✗ ||✔ || ✗ || ✗ || ✗
|✗
|}
<references/>

== Start parameters for Windows ==

On Windows, it is not possible to pass start parameters from the [https://www.chromium.org/developers Chromium documentation] to the myApps process.

== OS Settings for Windows ==
Windows settings can influence the display of ''Desktop notifications''. See [https://support.microsoft.com/en-us/help/4028678/windows-10-change-notification-settings Change notification settings in Windows 10/11] for details.

=== Windows 11 ===

* Windows 11 has a feature "do not disturb". This hides notifications if enabled.
* Windows 11 has a feature "focus". This enables "do not disturb" and thus hides notifications too.
* Windows 11 has priority settings for notifications. Ensure that VoIP notifications for calls are allowed any maybe also include myApps as an App which is allowed to show notifications.

== OS settings for Android ==
; Events : The appearance of notifications can be controlled here.

; Call accounts : For proper incoming call signaling, the call account ''myApps'' needs to be enabled. Note that on Samsung smartphones the call account switch likely toggles back and a few tries may need to be done until it persists. Please double-check the state.

; Preferred Calling Account : Choose which calling account (myApps/SIM/..) should be used for outgoing calls initiated from within the native phone app / phone book.

; Background data, unlimited data usage : Grant background data use to enable ''myApps'' to connect to the PBX immediately on an incoming call.

; Overlaying : This setting is not needed if call account ''myApps'' has been enabled. Should there be a reason for not enabling call account ''myApps'', the permission for overlaying needs to be granted on Android 10 or higher for proper call signaling.

Note: If no SIM card is installed some Android smartphones exhibit a problem dialing from the smartphone contacts. The contacts app shows a choice ''Select SIM card for this call'' but all possible dialers are greyed out. In this case make myApps the default phone app in Android settings ''Apps, Default apps, Telephony''.

== OS settings for iOS ==
; Notifications : The appearance of notifications can be controlled in iOS ''Settings, myApps''.

== OS settings for macOS ==

; Notifications : The appearance of notifications can be controlled in macOS ''Preferences, Notifications, myApps''.

=Troubleshooting=

myApps platform services can write various traces for debugging. Trace can be turned on and off selectively in the [[#Advanced settings|Advanced settings]].

The following trace flags can be set:

(''Recommended trace options are: '''App, Browser, ICE, TURN, Signaling and Audio'''. Please do not activate other flags unless innovaphone support says otherwise'')

{|
!style="text-align: left; font-weight: bold" | Abbreviation

!style="text-align: left; font-weight: bold" |code

!style="text-align: left; font-weight: bold" | description

|-
| App||0x000000001|| logs from the App Service itself
|-
| DNS||0x000000008|| logs DNS requests and results
|-
| HTTP client||0x000000080|| http client logs
|-
| TLS||0x000000400|| TLS logs
|-
| TCP||0x000000800|| TCP logs
|-
| LDS||0x000001000|| local domain sockets
|-
| WebSocket client||0x000004000|| logs outgoing websocket connections
|-
| App WebSocket||0x000008000|| logs app websocket connections (e.g. from PBX objects to an App Service or from the UI to the App Service)
|-
| UDP||0x000200000|| UDP logs
|-
| DTLS||0x000400000|| logs DTLS handshake and messages
|-
| Media||0x000800000|| logs media events
|-
| Media channel||0x001000000|| logs RTP/SCTP media connections
|-
| ICE||0x002000000|| logs ICE messages between peers
|-
| TURN||0x004000000|| logs TURN messages between peers
|-
| AppSharing||0x008000000|| logs AppSharing connection
|-
| Audio||0x010000000|| logs Audio connection and headset events
|-
| Video||0x020000000|| logs video connection and webcam events
|-
| Browser||0x040000000|| logs Chromium events
|-
| AppProxy||0x080000000|| logs requests which are proxied between the local webserver and the remote server
|-
| Webserver ||0x200000000|| enables webserver specific logs
|-
| Browser Console ||0x400000000|| logs browser console events
|-
| Signaling||0x800000000|| enables logs in the signaling module for debugging calls
|}
''code'' can be or'ed and used as value for the ''Log flags'' field in [[{{NAMESPACE}}:PBX/Config/myApps#Client_Settings|PBX/Config/myApps/Client Settings]].

== Windows ==
On Windows, traces are written to the <code>%LOCALAPPDATA%\innovaphone\myApps</code> directory. If you start myApps with --log-size as parameter, you can define the maximum size of a log file (e.g. --log-size=100000000 would be 100MB for each file)

:* myApps-''date-time''.txt : main log file for the platform services

:* myAppsOutlookSearch-''date-time''.txt : log file for the Outlook phone book access

:* myAppsHookController-''date-time''.txt : log file for the hot-key interceptor (see [[#Hot keys|Hot keys]])

; :myApps update installation traces are written to the <code>%windir%\temp\</code> directory.
:* myAppsInstall.txt: MSI installation file

; :myApps update service traces are written to the <code>%ProgramData%\innovaphone\myAppsUpdateService</code> directory.
:* myAppsUpdateService-''date-time''.txt: myApps update service traces

==Android==
traces can be sent by e-mail.

Also, an Android device might also be connected to a PC via an USB cable to get the traces. The files can be found in <code>Android/data/com.innovaphone.clientandroid/files</code>

== iOS ==
traces can be sent by e-mail.

== macOS ==
traces can be sent by e-mail.

Also, files can be found in <code>~/Library/Containers/com.innovaphone.client-ios/Data/Documents/</code>. Press ''Alt+N'' followed by space to get tilde ''~''.

== IP270 ==
Refer to the [[Reference16r1:Concept IP270#Troubleshooting|IP270 concept article section troubleshooting]].

= Known Problems =
[[:Category:Problem myApps platform services|Known Problems]]

= Related Articles =
* [[{{NAMESPACE}}:Concept_myApps]]
* [[{{NAMESPACE}}:Concept_myApps_Redundancy|Concept myApps Redundancy]]
* [[{{NAMESPACE}}:Concept_myApps_Office_Integration|Concept myApps Office Integration]]
* [[{{NAMESPACE}}:Concept_myAPPs_Search_in_local-Outlook_Contacts|Concept myAPPs Search in local Outlook Contacts]]
* [[{{NAMESPACE}}:Call_Detail_Record_CDR_PBX|Call Detail Records]]
* [[{{NAMESPACE}}:Concept Push Notifications for myPBX iOS and Android|Concept Push Notifications for myPBX iOS and Android]]
* [[Howto:Troubleshoot v13 Push with myApps for Android and iOS]]
* [[{{NAMESPACE}}:PBX/Config/myApps|Reference16r1:PBX/Config/myApps]]
* [[{{NAMESPACE}}:Concept_IP270|IP270 concept article]]
[[Category:Concept myApps platform services]]

Reference15r1:Concept App Service Fax

2026-06-02T12:41:38Z

Can: /* Known Issues */

[[Category:Concept|Apps]]

== Applies To ==
* innovaphone PBX from version 15r1

== Overview ==
The App Service Fax is an app service which can be installed on an innovaphone App Platform. It provides sending or receiving FAX documents with the innovaphone PBX and a user app to manage the documents.

== Features ==
* The App Service Fax sends PDF documents as FAX documents and converts received FAX documents to PDF.
* Cover page with user content for outgoing FAX documents.
* The Fax app can forward the received [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|fax documents by mail]], as notification only or with the document in PDF.
* Transmission reports and error notifications
* Contact search by using myApps search API providers, inclusive removing number decorations
* PBX node support
* Using several fax interfaces
* Group fax accounts
* Automatic deletion of older fax documents
* Customized email texts
* Mail2Fax
* [[{{NAMESPACE}}:Apps/PbxManager/App_Fax#Default_settings_for_mail_notifications|Centralized setup]] of mailing notifications
* App API (HTTP Post)

== Requirements ==
* innovaphone App Platform
* Device with a [[{{NAMESPACE}}:Gateway/Interfaces#FAX_interface|FAX interface]]
* One port license to register the fax interface towards the fax object
* A UC or fax license for each PBX user which is allowed to receive or send personal fax documents
* A fax (or UC) license for each group fax account

== Apps ==
'''Fax App (innovaphone-fax)''' 
This app is provided for the users to send fax documents and view or download received documents. The app reads the argument mailbox=<sip name> to work with the given account instead of the user logged in. It uses the com.innovaphone.search API to search for contacts with FAX numbers.

== Technical Overview ==
[[Image:Concept fax.png|concept_fax.png/]]

== Technical Concept ==
The innovaphone fax feature needs three parts:
* An innovaphone PBX with one or more fax objects
* One or more FAX interface registered to fax objects
* The Fax app on the App Platform (AP)

The app service on the AP provides a user app. Users can upload and download PDF documents. The app service converts them from and to a fax protocol compatible file (SFF).
The app service controls calls between the FAX interface and an external remote party. To do this, it requires a websocket connection to the PBX. The FAX interface accesses the file with authenticated WebDAV.
The documents saved in the app service are available with WebDAV with the app instance name as user name and the app instance password as password and in the directories
* <app-web-path>/sff for the SFF files
* <app-web-path>/doc for the PDF files

==== Additional Information ====
* Files of deleted fax jobs are permanently deleted after two weeks.
* The myApps background picture (myapps.png) is also included in all HTML mail bodies with customized mail texts, if these texts contain the string "url(cid:myapps.png)".

== Configuration ==

==== PBX Manager Plugin ====
All possible configurations can be done with the [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|PBX Manager plugin]] of the Fax App. The app service options can be set as well as the corresponding app objects in the PBX can be added, modified or deleted. Additionally, available devices with a fax interface can be found and configured. 
As with any app, the Fax App needs to be assigned to users and [[#Requirements| licensed accordingly]].

==== Mail Configuration (SMTP Server) ====
The fax service has a built-in SMTP server, which is needed for Mail2Fax. Here, you activate the mail reception for the fax service, enter the fax domain and configure the credentials for the authentication against the SMTP server. The mails can contain a subject and body for the cover page, and '''one PDF document can be appended as attachment'''. No cover page is added if both the subject and the mail content are empty. The destination number must be included in the recipient mail address in this format: <destination number>@<fax server domain>. The fax server domain is the app service domain or the configured domain if different. Recipient addresses do not match are discarded.

The supported charsets are UTF-8 and ISO 8859-1 (Latin 1).

As of 14r2 the SMTP server only listens on port 25. STARTTLS is possible.

For the Fax service to receive the mails in the first place, the mailserver needs to forward mails for the fax-domain to the APs IP address.

An example for an Exchange configuration can be found [[Howto14r2:Fax App - Mail2Fax with Exchange 2019|here]]

Starting with 16r1 you also can configure OAuth2 Authentications. You can have a look into our HowTo Article for assistance: [[Howto16r1:Configure OAuth2 E-Mail]]

==== Mail Configuration (SMTP Client) ====
The SMTP configuration is to be set in the Fax App PBX Manager plugin to make mail forwarding, transmission reports and notifications available. Only email addresses of the user configured in his PBX object are used and each user has to enable the several mails in the burger menu of the Fax app.
Available types of mails:
* Forwarding of a received document as PDF
* Incoming notifications without a document
* Error notifications for outgoing fax jobs
* Transmission confirmations
* Transmission reports for outgoing fax jobs as PDF

The mails are sent in the language the user set in myApps when the Fax app was last used, unless the language for mails was explicitly set within the app.

If a failure occurs and mails cannot be sent, the app service retries the mail transmission of a mail every 30 minutes, but no longer than two days.

==== PBX Node Configuration ====
The Fax app object can be assigned to a certain PBX node. If so, the node number is included within the user's fax number.

==== Group Fax Account ====
If a Group Fax app is configured, all users have the same group account with this app, and the same jobs. The mail addresses configured in this Group app are available and used instead of the user's mail addresses.

==== Sharing FAX Interfaces ====
FAX interfaces are normally registered to one app object. Other PBX Fax objects use these fax resources for calls if they are configured as external resource in the objects.

== App API ==
A document to be sent can be uploaded with the HTTP post command. The arguments within the HTTP URL sets the data of the new fax job. The job is created in the context of a user and is shown in the app. If the file is successfully saved, the job is directly queued for sending.

An API key must be configured to enable this feature and to authenticate the command.

Arguments of the HTTP post command in the URL:
* api-key: The configured authentication key. Mandatory.
* user-sip: The SIP of the user which sends the document. Mandatory.
* contact-name: An optional contact name shown in the app as contact (must be url-encoded).
* contact-number: The contact number used for sending the document. Mandatory.
* cover-page-subject: The subject of an optional cover page if should be created (must be url-encoded).
* cover-page-content: The content of an optional cover page if should be created (must be url-encoded).
* cover-page-content-html: If set to true, the cover page content is in HTML.
* app-object: The app object of the PBX used for sending. If not set, any is used.
* file-name: An optional file name of the uploaded file. Not used in the app.

Here an example:

curl -X POST "http://AP-DNS-name/DOMAIN/fax/?api-key=1234&user-sip=vgr&contact-number=00049703173009&cover-page-subject=Hello%20World&app-object=fax" -T testfax.pdf

== Troubleshooting ==
==== Log Files ====
The progress of fax jobs is reported in the log file:

Info job id 58, direction 0, progress 0, result 0

;direction
:0: incoming job
:1: outgoing job

;progress
:0: job created
:1: job queued
:2: job converted
:3: job finished

;result
:0: no error or successfully completed
:1: call aborted, repeating
:2: job stopped with an error
:3: job stopped caused by a conversion failure

==== App Service Log ====
*App
*App WebSocket
*SMTP (only if the problem is related to mail forwarding etc.)

==== App Service Logs for Mail2Fax ====
*App
*SMTP
*TLS
*TCP
*DB files

As these options generate extensive logs, please '''only enable them to track a current problem and disable them again afterwards'''.

== Known Issues ==
* '''Important:''' If using Mail2Fax, only '''ONE''' PDF attachment is allowed and can be handled by the Fax service. If multiple attachments are added the first PDF is used and all other attachments like more PDFs, pictures etc. are ignored.
* When using Mail2Fax the Display Name used for the Fax will be the FaxAPP Instance/GroupFax Instance configured in the user Object that have the matching email, if no FaxAPP Instances/GroupFax Instances are configured then a random one available in the system will be used for the Display Name.
* Since the Fax-Interface currently offers '''only''' support for 200 dpi and 400 dpi faxes, the same dpi is required on the remote party.
* Enabling of non-T.38 coder (audio-fax) is also possible for T.38-only capable devices.
: Has to be judged by the user himself, if selected fax interface features audio-fax.
* An IPVA is not capable of sending faxes with a non-T.38 coder (audio-fax coder, e.g. G.711A), because it has no DSPs which are needed for an audio-fax. Therefore, ''only'' T.38 faxes work on an IPVA FAX interface.
** '''Important:''' T.38 is to be enabled on ''all'' interfaces are used in the call flow, e.g. FAX, GW, SIP interface.
** '''Important:''' Often "exclusive" is activated on SIP interfaces. This prevents the renegotiation with the provider and T.38 can not be used. Therefore, the "exclusive" flag is to be disabled for such configurations.
* If the name (H.323) of an user is changed in the PBX, the data in the fax service are not accessible for this user and a new account is created for him in the service.
* With 14r1, faxes will be sent with 400dpi per default, and tried with a lower resolution again, if the remote side doesn't support 400dpi faxes. In a 13r3 fax app (no 400dpi support), you will see the first - non working - try, which is the normal behavior, since all faxes (working and non working) are shown in the fax app.
* Fax Fallback from T.38 to G.711 passthrough (AudioFax) is not supported by FaxAPP when a SIP Trunk with media-relay is used, which is often the case. The solution in such a scenario is to disable T.38 on the SIP Interface with media-relay and only use Audio-Fax.

== Related Articles ==
* For a more detailed description about the configuration, please refer to [[Courseware:IT_Plus_-_Fax|the book about fax application setup]] as part of our IT Plus Training.
* [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|PBX Manager plugin]] for the Fax App
* [[Howto: Mail2Fax with Exchange 2019]]

Especificações IP270

2026-05-19T13:06:03Z

Can: /* Hardware */

{{datasheet_header|innovaphone IP270|O telefone myapps}}
<div class="datasheets-mehrspaltig">
[[Image:IP270.png|ip270.png/]]
===Firmware===
Telefone compatível com a versão 16r1final ou superior
===Estrutura===
Ecrã tátil de 7 polegadas, 1280 x 800 pixeis

Teclado do telefone e teclado alfanumérico com exibição gradual

LED

9 teclas de função para controle de volume, rediscagem, mudo, controle de fone de ouvido e alto-falante

Base ajustável em quatro ângulos

Dimensões com base padrão (L x A x P): 260 x 200 x 160 mm

Peso: aprox. 1000 gramas com base padrão
===Interfaces===
1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) com “Power over Ethernet” de acordo com a norma 802.3af e “Energy Efficient Ethernet” de acordo com a norma 802.3az

1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) para conexão com PC

2 x porta USB 2.0 tipo A na parte traseira

2 x porta USB 2.0 tipo C na parte traseira

1 x porta USB 2.0 tipo C na parte traseira para fonte de alimentação externa

1 x porta USB 2.0 tipo C na parte frontal
===Hardware===
Alimentação: Power over Ethernet 802.3af, Classe 3 ou fonte de alimentação USB-C externa de 20 V, 6 W a máx. 2 A

2 GB de RAM LPDDR4, 8 GB de memória flash eMMC

Temperatura de operação: 0 °C a +45 °C

Temperatura de armazenamento: -10 °C a +70 °C

Umidade: 10% a 90% (sem condensação)

Modo de economia de energia com escurecimento automático do ecrã

Dois microfones, modo mãos-livres e escuta aberta

===Rede===
Suporte a IPv4/IPv6

SRTP (SDES, DTLS 1.2), ICE

STUN, TURN sobre UDP/TCP

Suporte a cliente DHCP

Suporte a NTP
===Administração===
* Administração centralizada por meio do aplicativo myApps-Devices para alterações de configuração e atualizações de firmware
* Geração de arquivos de rastreamento e exibição do registro de status
===Comunicação===
Aplicativo myApps-Linux-ClientSoftphone integrado para telefonia. Resumo dos recursos:
* Acesso com autenticação de dois fatores, bloqueio de tela
* Suporte a vários idiomas
* Suporte à conectividade de dispositivos USB (fone de ouvido, teclado, mouse, câmera)
* Serviços de cliente integrados (resumo):
** Aplicativos empresariais – Intranet social, gerenciamento de projetos, controle de horas
** Aplicativos de UC – Softphone, vídeo, chat, fax, correio de voz, conferência
** Agenda telefônica central de Contatos
* Suporte de áudio e vídeo
* Chamada em espera, sinalização respectiva de chamada em espera para o chamador
* Transferência de chamada, com/sem consulta
* Chamada em espera, com suporte a música de espera
* Lista de favoritos exibindo presença e estado da chamada
* Atendimento de chamadas a partir da lista de favoritos
* Desvio de chamadas: incondicional, ocupado e em caso de não atendimento
* Entrar / Sair de grupos para chamadas.
* Silenciamento de microfone
* Conferência a três, inclusive com assinantes externos
* Gravação de chamadas
* Vários toques/melodias integrados
===Licenças===
* São necessárias as Licenças do aplicativo de softphone (Número do pedido: 02-00050-004) e do aplicativo de telefone (Número do pedido: 02-00050-005) (uma vez por usuário)
* Para o registro de usuários do PBX, é necessária a Licença de porta do PBX (uma vez por usuário)
* Para o uso de aplicativos opcionais, podem ser necessárias licenças específicas de aplicativos (uma vez por usuário)
{{Template:CE-Decl_Phones|Declaração de conformidade|consulte a ficha informativa separada}}

===Código do produto: 01-00270-001===
</div>
{{Template:Datasheet_footer|PT}}
[[Category:Fichas técnicas Phones PT|{{PAGENAME}}]]

Especificações IP270

2026-05-19T13:05:54Z

Can: /* Hardware */

{{datasheet_header|innovaphone IP270|O telefone myapps}}
<div class="datasheets-mehrspaltig">
[[Image:IP270.png|ip270.png/]]
===Firmware===
Telefone compatível com a versão 16r1final ou superior
===Estrutura===
Ecrã tátil de 7 polegadas, 1280 x 800 pixeis

Teclado do telefone e teclado alfanumérico com exibição gradual

LED

9 teclas de função para controle de volume, rediscagem, mudo, controle de fone de ouvido e alto-falante

Base ajustável em quatro ângulos

Dimensões com base padrão (L x A x P): 260 x 200 x 160 mm

Peso: aprox. 1000 gramas com base padrão
===Interfaces===
1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) com “Power over Ethernet” de acordo com a norma 802.3af e “Energy Efficient Ethernet” de acordo com a norma 802.3az

1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) para conexão com PC

2 x porta USB 2.0 tipo A na parte traseira

2 x porta USB 2.0 tipo C na parte traseira

1 x porta USB 2.0 tipo C na parte traseira para fonte de alimentação externa

1 x porta USB 2.0 tipo C na parte frontal
===Hardware===
Alimentação: Power over Ethernet 802.3af, Classe 3 ou fonte de alimentação USB-C externa de 20 V, 6 W a máx. 2 A

2 GB de RAM LPDDR4, 8 GB de memória flash eMMC

Temperatura de operação: 0 °C a +45 °C

Temperatura de armazenamento: -10 °C a +70 °C

Umidade: 10% a 90% (sem condensação)

Modo de economia de energia com escurecimento automático da ecrã

Dois microfones, modo mãos-livres e escuta aberta

===Rede===
Suporte a IPv4/IPv6

SRTP (SDES, DTLS 1.2), ICE

STUN, TURN sobre UDP/TCP

Suporte a cliente DHCP

Suporte a NTP
===Administração===
* Administração centralizada por meio do aplicativo myApps-Devices para alterações de configuração e atualizações de firmware
* Geração de arquivos de rastreamento e exibição do registro de status
===Comunicação===
Aplicativo myApps-Linux-ClientSoftphone integrado para telefonia. Resumo dos recursos:
* Acesso com autenticação de dois fatores, bloqueio de tela
* Suporte a vários idiomas
* Suporte à conectividade de dispositivos USB (fone de ouvido, teclado, mouse, câmera)
* Serviços de cliente integrados (resumo):
** Aplicativos empresariais – Intranet social, gerenciamento de projetos, controle de horas
** Aplicativos de UC – Softphone, vídeo, chat, fax, correio de voz, conferência
** Agenda telefônica central de Contatos
* Suporte de áudio e vídeo
* Chamada em espera, sinalização respectiva de chamada em espera para o chamador
* Transferência de chamada, com/sem consulta
* Chamada em espera, com suporte a música de espera
* Lista de favoritos exibindo presença e estado da chamada
* Atendimento de chamadas a partir da lista de favoritos
* Desvio de chamadas: incondicional, ocupado e em caso de não atendimento
* Entrar / Sair de grupos para chamadas.
* Silenciamento de microfone
* Conferência a três, inclusive com assinantes externos
* Gravação de chamadas
* Vários toques/melodias integrados
===Licenças===
* São necessárias as Licenças do aplicativo de softphone (Número do pedido: 02-00050-004) e do aplicativo de telefone (Número do pedido: 02-00050-005) (uma vez por usuário)
* Para o registro de usuários do PBX, é necessária a Licença de porta do PBX (uma vez por usuário)
* Para o uso de aplicativos opcionais, podem ser necessárias licenças específicas de aplicativos (uma vez por usuário)
{{Template:CE-Decl_Phones|Declaração de conformidade|consulte a ficha informativa separada}}

===Código do produto: 01-00270-001===
</div>
{{Template:Datasheet_footer|PT}}
[[Category:Fichas técnicas Phones PT|{{PAGENAME}}]]

Especificações IP270

2026-05-19T13:05:41Z

Can: /* Licenças */

{{datasheet_header|innovaphone IP270|O telefone myapps}}
<div class="datasheets-mehrspaltig">
[[Image:IP270.png|ip270.png/]]
===Firmware===
Telefone compatível com a versão 16r1final ou superior
===Estrutura===
Ecrã tátil de 7 polegadas, 1280 x 800 pixeis

Teclado do telefone e teclado alfanumérico com exibição gradual

LED

9 teclas de função para controle de volume, rediscagem, mudo, controle de fone de ouvido e alto-falante

Base ajustável em quatro ângulos

Dimensões com base padrão (L x A x P): 260 x 200 x 160 mm

Peso: aprox. 1000 gramas com base padrão
===Interfaces===
1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) com “Power over Ethernet” de acordo com a norma 802.3af e “Energy Efficient Ethernet” de acordo com a norma 802.3az

1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) para conexão com PC

2 x porta USB 2.0 tipo A na parte traseira

2 x porta USB 2.0 tipo C na parte traseira

1 x porta USB 2.0 tipo C na parte traseira para fonte de alimentação externa

1 x porta USB 2.0 tipo C na parte frontal
===Hardware===
Alimentação: Power over Ethernet 802.3af, Classe 3 ou fonte de alimentação USB-C externa de 20 V, 6 W a máx. 2 A

2 GB de RAM LPDDR4, 8 GB de memória flash eMMC

Temperatura de operação: 0 °C a +45 °C

Temperatura de armazenamento: -10 °C a +70 °C

Umidade: 10% a 90% (sem condensação)

Modo de economia de energia com escurecimento automático da tela

Dois microfones, modo mãos-livres e escuta aberta
===Rede===
Suporte a IPv4/IPv6

SRTP (SDES, DTLS 1.2), ICE

STUN, TURN sobre UDP/TCP

Suporte a cliente DHCP

Suporte a NTP
===Administração===
* Administração centralizada por meio do aplicativo myApps-Devices para alterações de configuração e atualizações de firmware
* Geração de arquivos de rastreamento e exibição do registro de status
===Comunicação===
Aplicativo myApps-Linux-ClientSoftphone integrado para telefonia. Resumo dos recursos:
* Acesso com autenticação de dois fatores, bloqueio de tela
* Suporte a vários idiomas
* Suporte à conectividade de dispositivos USB (fone de ouvido, teclado, mouse, câmera)
* Serviços de cliente integrados (resumo):
** Aplicativos empresariais – Intranet social, gerenciamento de projetos, controle de horas
** Aplicativos de UC – Softphone, vídeo, chat, fax, correio de voz, conferência
** Agenda telefônica central de Contatos
* Suporte de áudio e vídeo
* Chamada em espera, sinalização respectiva de chamada em espera para o chamador
* Transferência de chamada, com/sem consulta
* Chamada em espera, com suporte a música de espera
* Lista de favoritos exibindo presença e estado da chamada
* Atendimento de chamadas a partir da lista de favoritos
* Desvio de chamadas: incondicional, ocupado e em caso de não atendimento
* Entrar / Sair de grupos para chamadas.
* Silenciamento de microfone
* Conferência a três, inclusive com assinantes externos
* Gravação de chamadas
* Vários toques/melodias integrados
===Licenças===
* São necessárias as Licenças do aplicativo de softphone (Número do pedido: 02-00050-004) e do aplicativo de telefone (Número do pedido: 02-00050-005) (uma vez por usuário)
* Para o registro de usuários do PBX, é necessária a Licença de porta do PBX (uma vez por usuário)
* Para o uso de aplicativos opcionais, podem ser necessárias licenças específicas de aplicativos (uma vez por usuário)
{{Template:CE-Decl_Phones|Declaração de conformidade|consulte a ficha informativa separada}}

===Código do produto: 01-00270-001===
</div>
{{Template:Datasheet_footer|PT}}
[[Category:Fichas técnicas Phones PT|{{PAGENAME}}]]

Especificações IP270

2026-05-18T15:15:49Z

Can: /* Carcaça */

{{datasheet_header|innovaphone IP270|O telefone myapps}}
<div class="datasheets-mehrspaltig">
[[Image:IP270.png|ip270.png/]]
===Firmware===
Telefone compatível com a versão 16r1final ou superior
===Estrutura===
Ecrã tátil de 7 polegadas, 1280 x 800 pixeis

Teclado do telefone e teclado alfanumérico com exibição gradual

LED

9 teclas de função para controle de volume, rediscagem, mudo, controle de fone de ouvido e alto-falante

Base ajustável em quatro ângulos

Dimensões com base padrão (L x A x P): 260 x 200 x 160 mm

Peso: aprox. 1000 gramas com base padrão
===Interfaces===
1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) com “Power over Ethernet” de acordo com a norma 802.3af e “Energy Efficient Ethernet” de acordo com a norma 802.3az

1 x Gigabit Ethernet: 1000-BASE-T (negociação automática), RJ-45 (conector modular 8P8C) para conexão com PC

2 x porta USB 2.0 tipo A na parte traseira

2 x porta USB 2.0 tipo C na parte traseira

1 x porta USB 2.0 tipo C na parte traseira para fonte de alimentação externa

1 x porta USB 2.0 tipo C na parte frontal
===Hardware===
Alimentação: Power over Ethernet 802.3af, Classe 3 ou fonte de alimentação USB-C externa de 20 V, 6 W a máx. 2 A

2 GB de RAM LPDDR4, 8 GB de memória flash eMMC

Temperatura de operação: 0 °C a +45 °C

Temperatura de armazenamento: -10 °C a +70 °C

Umidade: 10% a 90% (sem condensação)

Modo de economia de energia com escurecimento automático da tela

Dois microfones, modo mãos-livres e escuta aberta
===Rede===
Suporte a IPv4/IPv6

SRTP (SDES, DTLS 1.2), ICE

STUN, TURN sobre UDP/TCP

Suporte a cliente DHCP

Suporte a NTP
===Administração===
* Administração centralizada por meio do aplicativo myApps-Devices para alterações de configuração e atualizações de firmware
* Geração de arquivos de rastreamento e exibição do registro de status
===Comunicação===
Aplicativo myApps-Linux-ClientSoftphone integrado para telefonia. Resumo dos recursos:
* Acesso com autenticação de dois fatores, bloqueio de tela
* Suporte a vários idiomas
* Suporte à conectividade de dispositivos USB (fone de ouvido, teclado, mouse, câmera)
* Serviços de cliente integrados (resumo):
** Aplicativos empresariais – Intranet social, gerenciamento de projetos, controle de horas
** Aplicativos de UC – Softphone, vídeo, chat, fax, correio de voz, conferência
** Agenda telefônica central de Contatos
* Suporte de áudio e vídeo
* Chamada em espera, sinalização respectiva de chamada em espera para o chamador
* Transferência de chamada, com/sem consulta
* Chamada em espera, com suporte a música de espera
* Lista de favoritos exibindo presença e estado da chamada
* Atendimento de chamadas a partir da lista de favoritos
* Desvio de chamadas: incondicional, ocupado e em caso de não atendimento
* Entrar / Sair de grupos para chamadas.
* Silenciamento de microfone
* Conferência a três, inclusive com assinantes externos
* Gravação de chamadas
* Vários toques/melodias integrados
===Licenças===
* São necessárias a Licença do aplicativo de softphone (Número do pedido: 02-00050-004) e a Licença do aplicativo de telefone (Número do pedido: 02-00050-005) (uma vez por usuário)
* Para o registro de usuários do PBX, é necessária a Licença de porta do PBX (uma vez por usuário)
* Para o uso de aplicativos opcionais, podem ser necessárias licenças específicas de aplicativos (uma vez por usuário)
{{Template:CE-Decl_Phones|Declaração de conformidade|consulte a ficha informativa separada}}
===Código do produto: 01-00270-001===
</div>
{{Template:Datasheet_footer|PT}}
[[Category:Fichas técnicas Phones PT|{{PAGENAME}}]]

Howto:DE - Firstcomeurope - Universe SIP Trunk SIP-Provider (2026)

2026-04-22T14:22:09Z

Can: /* Remarks */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.universe-cloud.de/sip-trunk-connect|productname=Universe_SIP_Trunk|providername=Firstcomeurope}}
=== Remarks ===
* This profile replaces DE - CITEC_AG - Universe_SIP_Trunk in v16r1.

* CLNS only works if CGPN number sent has 12 digits(the 00 does not count), if shorter the call will be displayed as anonymous

* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_yes}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

==Configuration==
Use profile ''DE-Firstcomeurope-Universe_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Firstcomeurope - Universe SIP Trunk SIP-Provider (2026)

2026-04-22T14:20:58Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.universe-cloud.de/sip-trunk-connect|productname=Universe_SIP_Trunk|providername=Firstcomeurope}}
=== Remarks ===
* CLNS only works if CGPN number sent has 12 digits(the 00 does not count), if shorter the call will be displayed as anonymous

* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_yes}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

==Configuration==
Use profile ''DE-Firstcomeurope-Universe_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Firstcomeurope - Universe SIP Trunk SIP-Provider (2026)

2026-04-22T14:14:58Z

Can: /* Remarks */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https:|productname=Universe_SIP_Trunk|providername=Firstcomeurope}}
=== Remarks ===
* This profile replaces DE-CITEC_AG-Universe_SIP_Trunk in v16r1.

* CLNS only works if CGPN number sent has 12 digits(the 00 does not count), if shorter the call will be displayed as anonymous

* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_yes}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

==Configuration==
Use profile ''DE-Firstcomeurope-Universe_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Firstcomeurope - Universe SIP Trunk SIP-Provider (2026)

2026-04-22T14:07:03Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https:|productname=Universe_SIP_Trunk|providername=Firstcomeurope}}
=== Remarks ===
* This profile replaces DE-CITEC_AG-Universe_SIP_Trunk in v16r1, if you are migrating from v15r1 to v16r1 is necessary to re-configure the SIP trunk with the new SIP Profile.

* CLNS only works if CGPN number sent has 12 digits(the 00 does not count), if shorter the call will be displayed as anonymous

* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_yes}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

==Configuration==
Use profile ''DE-Firstcomeurope-Universe_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Firstcomeurope - Universe SIP Trunk SIP-Provider (2026)

2026-04-22T14:05:31Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https:|productname=Universe_SIP_Trunk|providername=Firstcomeurope}}
=== Remarks ===
* CLNS only works if CGPN number sent has 12 digits(the 00 does not count), if shorter the call will be displayed as anonymous

* Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects

* 'Set Calling = Diverting No' recommended in PBX 'Trunk' objects

* Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_yes}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

==Configuration==
Use profile ''DE-Firstcomeurope-Universe_SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - htp - SIP Trunk pure SIP-Provider (2026)

2026-04-22T11:39:53Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.htp.net/telefon-internet/sip-einzelanschluss|productname=SIP_Trunk_pure|providername=htp}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A and G711U

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-htp-SIP_Trunk_pure'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' DigiCert Global Root G2' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - htp - SIP Trunk pure SIP-Provider (2026)

2026-04-22T11:33:26Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https:|productname=SIP_Trunk_pure|providername=htp}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}
; XFER CONS ALERT : {{SIP_TEST_FACT_XFER CONS ALERT}}
; XFER CONS : {{SIP_TEST_FACT_XFER CONS}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A and G711U

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer :
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consconn}}
: {{Template:SIP_Profile_Test_CALL_TRANSFER_consalert}}

==Configuration==
Use profile ''DE-htp-SIP_Trunk_pure'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' DigiCert Global Root G2' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:CH - Winet - SIP Trunk SIP-Provider (2026)

2026-04-22T11:28:31Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.winet.ch/geschaeftskunden/ucc-voice/rufnummern-schweiz/|productname=SIP_Trunk|providername=Winet}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SDP ICE : {{SIP_TEST_FACT_SDP ICE}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U, G722, G729, OPUS_NB and OPUS_WB
: supported onnet (VoIP to VoIP): G711A and G711U

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''CH-Winet-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' SwissSign Gold CA - G2' certificate to the trust list of your SBC</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:AT - Innosoft - SIP Trunk SIP-Provider (2026)

2026-04-22T11:22:30Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.innosoft.at/festnetz/business/sip-trunk-fuer-voip-telefonanlage|productname=SIP_Trunk|providername=Innosoft}}

=== Remarks ===
- The CLIR test shows us that SIP Provider sends "Anonymous" in the From Header but there is no Privacy ID header included, which is why the test failed.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLIR : {{SIP_TEST_FACT_CLIR}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_no}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_strange}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''AT-Innosoft-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Template:SIP TEST V16 HINT

2026-04-22T11:19:15Z

Can: Created page with "A most recent v16r1 firmware is required to use this SIP-profile. For hints regarding upgrade to v16r1, see Howto16r1:Firmware_Upgrade_V15r1_V16r1 New profiles are added in the course of our V16R1 software Service Releases, see ReleaseNotes16r1:Firmware. Here is an ''up to date list of tested SIP providers''."

A most recent v16r1 firmware is required to use this SIP-profile. For hints regarding upgrade to v16r1, see [[Howto16r1:Firmware_Upgrade_V15r1_V16r1]]

New profiles are added in the course of our V16R1 software Service Releases, see [[ReleaseNotes16r1:Firmware]]. Here is an ''[[:Category:Automated_SIP_Tests|up to date list of tested SIP providers]]''.

Howto:AT - Innosoft - SIP Trunk SIP-Provider (2026)

2026-04-22T11:17:58Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 22th, 2026|url=https://www.innosoft.at/festnetz/business/sip-trunk-fuer-voip-telefonanlage|productname=SIP_Trunk|providername=Innosoft}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLIR : {{SIP_TEST_FACT_CLIR}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_no}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_strange}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''AT-Innosoft-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>

: {{SIP_TEST_V16_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-04-21T13:35:20Z

Can: /* Remarks */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 21st, 2026|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

* Important update made on SIP Profile in version '''v15r1sr11''' that includes new SIP advance tweaks: ''/reg-redirect-support /prefer-plain-auth /contact-uri-without-port'' . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-04-21T13:34:50Z

Can: /* Configuration */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=April 21st, 2026|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

* Important update made on SIP Profile in version '''v15r1sr11''' that includes new SIP advance tweaks: ''/reg-redirect-support /prefer-plain-auth /contact-uri-without-port /session-expires 900'' . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TLS SIP-Provider (2025)

2026-02-13T10:54:38Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=September 4th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.
* Important update made on SIP Profile in version '''v15r1sr11''' that includes new SIP advance tweaks: ''/reg-redirect-support /prefer-plain-auth /contact-uri-without-port /session-expires 900'' . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' rm1.dtmd4sbc801.sipt.vodafone.de' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-02-13T10:53:52Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 23th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.
* Important update made on SIP Profile in version '''v15r1sr11''' that includes new SIP advance tweaks: ''/reg-redirect-support /prefer-plain-auth /contact-uri-without-port /session-expires 900'' . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-02-13T10:53:39Z

Can:

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-02-13T10:53:21Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 23th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.
* Important update made on SIP Profile in version '''v15r1sr11''' that includes new SIP advance tweaks: /reg-redirect-support /prefer-plain-auth /contact-uri-without-port /session-expires 900 . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2026-02-13T10:53:03Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 23th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.
* Important update made on SIP Profile in version v15r1sr11 that includes new SIP advance tweaks: /reg-redirect-support /prefer-plain-auth /contact-uri-without-port /session-expires 900 . So after you update to v15r1sr11 or higher go to SIP Profile and click "OK" button to refresh the settings, if you are not using it yet you can manually set this settings in Expert Mode in the SIP Advance Field to avoid registraiton issues.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - KEVAG Telekom - SIP Trunk SIP-Provider (2025)

2025-12-02T09:18:24Z

Can: /* Configuration */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=December 2nd, 2025|url=https://www.kevag-telekom.de/tarife|productname=SIP_Trunk|providername=KEVAG_Telekom}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_strange}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-KEVAG_Telekom-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - KEVAG Telekom - SIP Trunk SIP-Provider (2025)

2025-12-02T09:18:11Z

Can: Created page with "== Summary == {{Template:SIP_TEST_STATUS_complete|update=December 2nd, 2025|url=https://www.kevag-telekom.de/tarife|productname=SIP_Trunk|providername=KEVAG_Telekom}} === {{SIP_TEST_ISSUES_MR_TITLE}} === {{SIP_TEST_ISSUES_MR_INTRO}} ; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}} ; FAX T38 : {{SIP_TEST_FACT_FAX T38}} ; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} ; MOBILITY : {{SIP_TEST_FACT_MOBILITY}} ; SIP INFO : {{SIP_TEST_FACT_SIP INFO}} <sma..."

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=December 2nd, 2025|url=https://www.kevag-telekom.de/tarife|productname=SIP_Trunk|providername=KEVAG_Telekom}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_strange}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-KEVAG_Telekom-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V12_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:USA - Telnyx - SIP Trunk SIP-Provider (2025)

2025-11-06T11:18:08Z

Can: Can moved page Howto:USA - Telnyx - SIP Trunk SIP-Provider (2020) to Howto:USA - Telnyx - SIP Trunk SIP-Provider (2025) without leaving a redirect: update of date

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=November 6th, 2025|url=https://telnyx.com/products/sip-trunking|productname=SIP_Trunk|providername=Telnyx}}
=== Remarks ===
Telnyx offers a location SIP Server option for better connectivity in Europe with address sip.telnyx.eu, this is the default SIP Server configured by the Profile, if you are located in America you can manually change to sip.telnyx.com while inserting the account details.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 ONNET : {{SIP_TEST_FACT_FAX T38 ONNET}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_yes_FAILOVER_no|timeout=2 minutes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_no_clns_but_clns_onnet_OK}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_history_or_diversion}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}} {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC|codec="G711A"}}
: {{Template:SIP_Profile_Test_T38_PSTN_yes_onnet_no_fallback_no}}

; Codecs : supported to/from PSTN: G711A, G711U and G722
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_no}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''USA-Telnyx-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>FAX requires exclusive G711A codec</nowiki>
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' USERTrust RSA Certification Authority' certificate to the trust list of your SBC</nowiki>
* <nowiki>Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:USA - Telnyx - SIP Trunk SIP-Provider (2025)

2025-11-06T11:16:56Z

Can:

Reference15r1:Concept App Service Fax

2025-10-24T10:50:28Z

Can: /* Known Issues */

[[Category:Concept|Apps]]

== Applies To ==
* innovaphone PBX from version 15r1

== Overview ==
The App Service Fax is an app service which can be installed on an innovaphone App Platform. It provides sending or receiving FAX documents with the innovaphone PBX and a user app to manage the documents.

== Features ==
* The App Service Fax sends PDF documents as FAX documents and converts received FAX documents to PDF.
* Cover page with user content for outgoing FAX documents.
* The Fax app can forward the received [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|fax documents by mail]], as notification only or with the document in PDF.
* Transmission reports and error notifications
* Contact search by using myApps search API providers, inclusive removing number decorations
* PBX node support
* Using several fax interfaces
* Group fax accounts
* Automatic deletion of older fax documents
* Customized email texts
* Mail2Fax
* [[{{NAMESPACE}}:Apps/PbxManager/App_Fax#Default_settings_for_mail_notifications|Centralized setup]] of mailing notifications
* App API (HTTP Post)

== Requirements ==
* innovaphone App Platform
* Device with a [[{{NAMESPACE}}:Gateway/Interfaces#FAX_interface|FAX interface]]
* One port license to register the fax interface towards the fax object
* A UC or fax license for each PBX user which is allowed to receive or send personal fax documents
* A fax (or UC) license for each group fax account

== Apps ==
'''Fax App (innovaphone-fax)''' 
This app is provided for the users to send fax documents and view or download received documents. The app reads the argument mailbox=<sip name> to work with the given account instead of the user logged in. It uses the com.innovaphone.search API to search for contacts with FAX numbers.

== Technical Overview ==
[[Image:Concept fax.png|concept_fax.png/]]

== Technical Concept ==
The innovaphone fax feature needs three parts:
* An innovaphone PBX with one or more fax objects
* One or more FAX interface registered to fax objects
* The Fax app on the App Platform (AP)

The app service on the AP provides a user app. Users can upload and download PDF documents. The app service converts them from and to a fax protocol compatible file (SFF).
The app service controls calls between the FAX interface and an external remote party. To do this, it requires a websocket connection to the PBX. The FAX interface accesses the file with authenticated WebDAV.
The documents saved in the app service are available with WebDAV with the app instance name as user name and the app instance password as password and in the directories
* <app-web-path>/sff for the SFF files
* <app-web-path>/doc for the PDF files

==== Additional Information ====
* Files of deleted fax jobs are permanently deleted after two weeks.
* The myApps background picture (myapps.png) is also included in all HTML mail bodies with customized mail texts, if these texts contain the string "url(cid:myapps.png)".

== Configuration ==

==== PBX Manager Plugin ====
All possible configurations can be done with the [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|PBX Manager plugin]] of the Fax App. The app service options can be set as well as the corresponding app objects in the PBX can be added, modified or deleted. Additionally, available devices with a fax interface can be found and configured. 
As with any app, the Fax App needs to be assigned to users and [[#Requirements| licensed accordingly]].

==== Mail Configuration (SMTP Server) ====
The fax service has a built-in SMTP server, which is needed for Mail2Fax. Here, you activate the mail reception for the fax service, enter the fax domain and configure the credentials for the authentication against the SMTP server. The mails can contain a subject and body for the cover page, and '''one PDF document can be appended as attachment'''. No cover page is added if both the subject and the mail content are empty. The destination number must be included in the recipient mail address in this format: <destination number>@<fax server domain>. The fax server domain is the app service domain or the configured domain if different. Recipient addresses do not match are discarded.

The supported charsets are UTF-8 and ISO 8859-1 (Latin 1).

As of 14r2 the SMTP server only listens on port 25. STARTTLS is possible.

For the Fax service to receive the mails in the first place, the mailserver needs to forward mails for the fax-domain to the APs IP address.

An example for an Exchange configuration can be found [[Howto14r2:Fax App - Mail2Fax with Exchange 2019|here]]

Starting with 16r1 you also can configure OAuth2 Authentications. You can have a look into our HowTo Article for assistance: [[Howto16r1:Configure OAuth2 E-Mail]]

==== Mail Configuration (SMTP Client) ====
The SMTP configuration is to be set in the Fax App PBX Manager plugin to make mail forwarding, transmission reports and notifications available. Only email addresses of the user configured in his PBX object are used and each user has to enable the several mails in the burger menu of the Fax app.
Available types of mails:
* Forwarding of a received document as PDF
* Incoming notifications without a document
* Error notifications for outgoing fax jobs
* Transmission confirmations
* Transmission reports for outgoing fax jobs as PDF

The mails are sent in the language the user set in myApps when the Fax app was last used, unless the language for mails was explicitly set within the app.

If a failure occurs and mails cannot be sent, the app service retries the mail transmission of a mail every 30 minutes, but no longer than two days.

==== PBX Node Configuration ====
The Fax app object can be assigned to a certain PBX node. If so, the node number is included within the user's fax number.

==== Group Fax Account ====
If a Group Fax app is configured, all users have the same group account with this app, and the same jobs. The mail addresses configured in this Group app are available and used instead of the user's mail addresses.

==== Sharing FAX Interfaces ====
FAX interfaces are normally registered to one app object. Other PBX Fax objects use these fax resources for calls if they are configured as external resource in the objects.

== App API ==
A document to be sent can be uploaded with the HTTP post command. The arguments within the HTTP URL sets the data of the new fax job. The job is created in the context of a user and is shown in the app. If the file is successfully saved, the job is directly queued for sending.

An API key must be configured to enable this feature and to authenticate the command.

Arguments of the HTTP post command in the URL:
* api-key: The configured authentication key. Mandatory.
* user-sip: The SIP of the user which sends the document. Mandatory.
* contact-name: An optional contact name shown in the app as contact (must be url-encoded).
* contact-number: The contact number used for sending the document. Mandatory.
* cover-page-subject: The subject of an optional cover page if should be created (must be url-encoded).
* cover-page-content: The content of an optional cover page if should be created (must be url-encoded).
* cover-page-content-html: If set to true, the cover page content is in HTML.
* app-object: The app object of the PBX used for sending. If not set, any is used.
* file-name: An optional file name of the uploaded file. Not used in the app.

Here an example:

curl -X POST "http://AP-DNS-name/DOMAIN/fax/?api-key=1234&user-sip=vgr&contact-number=00049703173009&cover-page-subject=Hello%20World&app-object=fax" -T testfax.pdf

== Troubleshooting ==
==== Log Files ====
The progress of fax jobs is reported in the log file:

Info job id 58, direction 0, progress 0, result 0

;direction
:0: incoming job
:1: outgoing job

;progress
:0: job created
:1: job queued
:2: job converted
:3: job finished

;result
:0: no error or successfully completed
:1: call aborted, repeating
:2: job stopped with an error
:3: job stopped caused by a conversion failure

==== App Service Log ====
*App
*App WebSocket
*Browser
*SMTP (only if the problem is related to mail forwarding etc.)

==== App Service Logs for Mail2Fax ====
*App
*Smtp
*TLS
*TCP
*DB files
-> Since these options creates huge logs, please just enable for tracing a current problem and deactivate afterward

== Known Issues ==
* '''Important:''' If using Mail2Fax, only '''ONE''' PDF attachment is allowed and can be handled by the Fax service. If multiple attachments are added the first PDF is used and all other attachments like more PDFs, pictures etc. are ignored.
* Since the Fax-Interface currently offers '''only''' support for 200 dpi and 400 dpi faxes, the same dpi is required on the remote party.
* Enabling of non-T.38 coder (audio-fax) is also possible for T.38-only capable devices.
: Has to be judged by the user himself, if selected fax interface features audio-fax.
* An IPVA is not capable of sending faxes with a non-T.38 coder (audio-fax coder, e.g. G.711A), because it has no DSPs which are needed for an audio-fax. Therefore, ''only'' T.38 faxes work on an IPVA FAX interface.
** '''Important:''' T.38 is to be enabled on ''all'' interfaces are used in the call flow, e.g. FAX, GW, SIP interface.
** '''Important:''' Often "exclusive" is activated on SIP interfaces. This prevents the renegotiation with the provider and T.38 can not be used. Therefore, the "exclusive" flag is to be disabled for such configurations.
* If the name (H.323) of an user is changed in the PBX, the data in the fax service are not accessible for this user and a new account is created for him in the service.
* With 14r1, faxes will be sent with 400dpi per default, and tried with a lower resolution again, if the remote side doesn't support 400dpi faxes. In a 13r3 fax app (no 400dpi support), you will see the first - non working - try, which is the normal behavior, since all faxes (working and non working) are shown in the fax app.
* Fax Fallback from T.38 to G.711 passthrough (AudioFax) is not supported by FaxAPP when a SIP Trunk with media-relay is used, which is often the case. The solution in such a scenario is to disable T.38 on the SIP Interface with media-relay and only use Audio-Fax.

== Related Articles ==
* For a more detailed description about the configuration, please refer to [https://class.innovaphone.com/moodle2/course/view.php?id=1705&topic=0#section-2 the book about fax application setup] as part of our IT Plus Training.
* [[{{NAMESPACE}}:Apps/PbxManager/App_Fax|PBX Manager plugin]] for the Fax App
* [[Howto: Mail2Fax with Exchange 2019]]

Howto13r3:Step-by-Step Media Relay connection for third party phone

2025-09-09T14:36:53Z

Can: /* Configuration-check of the Reverse Proxy (RP) */

[[Category:Step-by-Step]]

{{HOWTOMOD13r3 Translation Info}}

This article defines the recommended configuration to connect third party phones also via network boundaries. The configuration ensures media connectivity and compatibility to WebRTC without relying on ICE and DTLS on the phone.

[[Image:Media-relay-endpoints-overview.png|media-relay-endpoints-overview.png/]]

== Purpose ==
A common problem is that third-party SIP phones do not support ICE or DTLS protocols. As a result audio connections across NAT boundaries are often a challenge. To solve this problem, you can enable Media Relay, but the Media Relay endpoint address must be a public IP address so that the external SIP phone can send its audio to this address.

Furthermore, WebRTC endpoints require ICE and DTLS to establish an audio connection. Since not all SIP phones support these features, enabling Media Relay for these 3rd party phones solves this issue.

=== Features ===
*Send Audio traffic across NAT boundaries without using the ICE mechanism.

== Requirements ==
*The firmware has to be at least 13r3
*innovaphone PBX
*3rd party SIP phone
*TURN Server
*External endpoints must be connected via the innovaphone Reverse Proxy

== Things to know before you begin ==
*The public IP address of the network (External IP of the Firewall or NAT Router)
*You must have access to the firewall or NAT router to be able to configure port forwardings
*You could route the RTP directly to the PBX, without using a TURN server. This is not recommended as it would allow attacks on your PBX.
*If your third party SIP Phone supports ICE this solution doesn't apply. No need to use Media Relay.

== Configuration ==
In this scenario, we are going to configure the above picture to allow an external third party SIP device to send its audio to an internal destination. This Wiki article does not explain how to register a SIP phone to an innovaphone PBX via a reverse proxy. If you need help to accomplish this, please read the instructions in our [[Courseware:IT_Advanced_-_07_Public_access_to_PBX_resources_(practice)|Advanced Training Part 2]] materials.

Please register your SIP device to your PBX via the reverse proxy.

=== Configuration on the User Object ===
Please set the option ''Media Relay'' on the hardware ID of the user object on which your SIP device is registered. Do not set the Media Relay option globally in the PBX (PBX/Config/General)! This option is no longer required as you can enable Media Relay for each hardware ID individually.

[[Image:Media-relay-endpoints-1.png|media-relay-endpoints-1.png/]]

=== Configuration of the PBX ===
Under PBX/Config/General of your Advanced UI you will find a configuration option called ''Media Relay Endpoints''. Leave the ''Firewall public IP'' configuration empty, as this option is only necessary if you would route the traffic directly to the PBX. As mentioned above, this is not recommended. Instead, use a TURN server that you would use for your myApps client anyway.

To enable the use of a TURN server you must to activate the checkbox right next to the ''Media Relay Endpoints'' configuration. This option allows the PBX to send the TURN server IP address as the "connection address" in the SDP for all devices that are registered via the reverse proxy and use Media Relay. (see first step)

[[Image:Media-relay-endpoints-2.png|media-relay-endpoints-2.png/]]

In order to send the correct (public) IP address, you need to configure the option ''TURN Public Address'' in your TURN server settings(IP4/NAT/General).

[[Image:Media-relay-endpoints-4.png|media-relay-endpoints-4.png/]]

=== Configuration-check of the Reverse Proxy (RP) ===
Although this step-by-step article is focussing on relaying the voice correctly for external SIP-phones, there is one extra config-check which must be considered for a good and lasting RTP-stream.

In the Advanced UI of the Reverse Proxy (RP) you can/must configure the '''Public NAT router address''' under Services/Reverse-Proxy.
Here you can configure the Public/External IP address of the Reverse Proxy if the Reverse Proxy is not acting/configured as NAT Router.
This config-field is used to adjust/change the 'Record-Route'-Headers in the SIP-signalling towards the external SIP-device, to force future requests in the dialog to be routed through the proxy.

[[Image:RP-Public_nat_router_address.png|rp-public_nat_router_address.png/]]

===RTP Range Configuration===
If you want to restrict connections to a specific port range, you can create an RTP port range on your TURN server. Keep in mind that this range will be used for every call from any device that is using this TURN server. This applies to both internal and external devices, so you should not restrict the number of ports too much. The RTP port range is configured on IP4/General/Settings of your TURN server. Set the ''First UDP-RTP Port'' and then the ''Number of Ports''.

[[Image:Media-relay-endpoints-3.png|media-relay-endpoints-3.png/]]

===Firewall Configuration===
You need to create port forwardings on your firewall. A port forwarding for your RTP/UDP ports must be configured towards the TURN server. E.g If you configured a RTP port Range 16384 to 32767, a port forwarding for those exact ports have to be configured on your firewall.

== Verification ==

Please call any extension number in your PBX. You should be able to hear and talk to the other party on the call.

== Known issues ==
By default, calling a Voicemail has no audio and while calling a conference there is also no audio after the PIN has been dialed. However there is a workaround by forcing these calls over an extra Gateway with Media-Relay activated.

== Known Limitations ==
Currently Video relay it's not supported with this setup, only Audio relay.

== Related Articles ==
*[[Reference13r3:Concept Third Party Phones]]

Howto:DE - Vodafone - IP Anlagen Anschluss TLS SIP-Provider (2025)

2025-09-04T12:43:44Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=September 4th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_yes}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' rm1.dtmd4sbc801.sipt.vodafone.de' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:FR - Peoplefone - SIP Trunk TLS SIP-Provider (2025)

2025-08-20T13:27:50Z

Can: Can moved page Howto:FR - Peoplefone - SIP Trunk TLS SIP-Provider (2024) to Howto:FR - Peoplefone - SIP Trunk TLS SIP-Provider (2025) without leaving a redirect: Misspelled title

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=August 20th, 2025|url=https://www.peoplefone.com/de-de/kunden/produkte/telefonie/peoplefone-sip-trunk?gad_source=1&gclid=EAIaIQobChMI8qLwuY-IhwMV8pCDBx1VzhK8EAAYASAAEgLdlPD_BwE|productname=SIP_Trunk_TLS|providername=Peoplefone}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; RALERT DISC : {{SIP_TEST_FACT_RALERT DISC}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_no}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}
:{{Template:SIP_Profile_Test_RALERT_DISC_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''FR-Peoplefone-SIP_Trunk_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' sips.peoplefone.ch' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:FR - Peoplefone - SIP Trunk TLS SIP-Provider (2025)

2025-08-20T13:26:46Z

Can: Created page with "== Summary == {{Template:SIP_TEST_STATUS_complete|update=August 20th, 2025|url=https://www.peoplefone.com/de-de/kunden/produkte/telefonie/peoplefone-sip-trunk?gad_source=1&gclid=EAIaIQobChMI8qLwuY-IhwMV8pCDBx1VzhK8EAAYASAAEgLdlPD_BwE|productname=SIP_Trunk_TLS|providername=Peoplefone}} === {{SIP_TEST_ISSUES_MR_TITLE}} === {{SIP_TEST_ISSUES_MR_INTRO}} ; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}} ; FAX T38 : {{SIP_TEST_FACT_FAX T38}} ; FAX T38ANDAUDIO : {{SIP_TEST_FACT_..."

Howto:IT - VoipVoice - SIP Trunk SIP-Provider (2018)

2025-08-19T13:33:26Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=May 15th, 2024|url=http://www.voipvoice.it/numeri-voip-cosa-sono.html|productname=Voip|providername=VoipVoice}}

=== Remarks ===
{{ Template:SIP_TEST_NO_NIGHTLY_TESTS | fw-version = 14r2 Service Release 1 (1420322)}}

=== {{SIP_TEST_ISSUES_NO_MR_TITLE}} ===
{{SIP_TEST_ISSUES_NO_MR_INTRO}}
; CLIR : {{SIP_TEST_FACT_CLIR}}
; CLNS ONNET : {{SIP_TEST_FACT_CLNS ONNET}}
; CLNS : {{SIP_TEST_FACT_CLNS}}
; FAX AUDIO : {{SIP_TEST_FACT_FAX AUDIO}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
: {{SIP_TEST_FACT__unreliable}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REDIR DIVHDR : {{SIP_TEST_FACT_REDIR DIVHDR}}
; REDIR HISTHDR : {{SIP_TEST_FACT_REDIR HISTHDR}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]]

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_ALTERNATE_INTRO}}
{{SIP_TEST_ISSUES_MR_INTRO}}
{{SIP_TEST_ISSUES_NO_ALTERNATE_ISSUES}}

== Test Results ==
{{SIP_TEST_TESTRESULT_BOTH_INTRO}}
=== {{SIP_TEST_RESULTS_NO_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} {{Template:SIP_Profile_Test_DTMF_unreliable}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_no}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}} {{Template:SIP_Profile_Test_DTMF_unreliable}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_no}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_no}} {{Template:SIP_Profile_Test_CLNS_REDIRECT_no_clns_at_all}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_no}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_clns_no_history_or_diversion}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

==Configuration==
Use profile ''IT-VoipVoice-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>STUN required on all endpoints if media-relay is not used</nowiki>
* <nowiki>'Set Calling = Diverting No' recommended in PBX 'Trunk' objects</nowiki>

: {{SIP_TEST_V12_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Kwebbl - SIP Trunk SIP-Provider (2025)

2025-08-19T11:02:10Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 16th, 2025|url=https://kwebbl.com/it_distributors|productname=SIP_Trunk|providername=Kwebbl}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; 180 RINGING : {{SIP_TEST_FACT_180 RINGING}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; REDIR 302 : {{SIP_TEST_FACT_REDIR 302}}
; REVERSE MEDIA : {{SIP_TEST_FACT_REVERSE MEDIA}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_UDP_TCP_TLS}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_no}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_yes}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_no}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}} {{Template:SIP_Profile_Test_MobilityCall_no_ringing}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Kwebbl-SIP_Trunk'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>Alert not signalled, 'Carrier w/o Alerting' required in all PBX 'Mobility' objects</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TLS SIP-Provider (2025)

2025-07-25T10:20:06Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 25th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' rm1.dtmd4sbc801.sipt.vodafone.de' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TLS SIP-Provider (2025)

2025-07-25T10:19:11Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 25th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html //|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' rm1.dtmd4sbc801.sipt.vodafone.de' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TLS SIP-Provider (2025)

2025-07-25T10:17:18Z

Can: Created page with "== Summary == {{Template:SIP_TEST_STATUS_ongoing|update=July 25th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html //|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}} === {{SIP_TEST_ISSUES_MR_TITLE}} === {{SIP_TEST_ISSUES_MR_INTRO}} ; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}} ; FAX T38 : {{SIP_TEST_FACT_FAX T38}} ; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}} ; MOBILITY : {{SIP_TEST_FACT_MOBILI..."

== Summary ==
{{Template:SIP_TEST_STATUS_ongoing|update=July 25th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html //|productname=IP_Anlagen_Anschluss_TLS|providername=Vodafone}}

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SDP RTCP MUX : {{SIP_TEST_FACT_SDP RTCP MUX}}
; SDP VIDEO : {{SIP_TEST_FACT_SDP VIDEO}}
: {{SIP_TEST_FACT__unreliable}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{SIP_Profile_Test_Registration_TLS_NO_UDP_NO_TCP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TLS'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.

Please note the following configuration hints:
* <nowiki>If you intend to use SIPS (SIP/TLS) registration, you need to add the ' rm1.dtmd4sbc801.sipt.vodafone.de' certificate to the trust list of your SBC</nowiki>

: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2025-07-23T15:48:03Z

Can: /* Configuration */

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 23th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V15_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]

Howto:DE - Vodafone - IP Anlagen Anschluss TCP SIP-Provider (2023)

2025-07-23T15:47:51Z

Can:

== Summary ==
{{Template:SIP_TEST_STATUS_complete|update=July 23th, 2025|url=https://www.vodafone.de/business/festnetz-internet/ip-anlagen-anschluss.html|productname=IP_Anlagen_Anschluss_TCP|providername=Vodafone}}
=== Remarks ===
* Tests made with the IP Anlagen Anschluss R6 version.

=== {{SIP_TEST_ISSUES_MR_TITLE}} ===
{{SIP_TEST_ISSUES_MR_INTRO}}
; EARLY MEDIA INBOUND : {{SIP_TEST_FACT_EARLY MEDIA INBOUND}}
; FAX T38 : {{SIP_TEST_FACT_FAX T38}}
; FAX T38ANDAUDIO : {{SIP_TEST_FACT_FAX T38ANDAUDIO}}
; MOBILITY : {{SIP_TEST_FACT_MOBILITY}}
; SIP INFO : {{SIP_TEST_FACT_SIP INFO}}

{{SIP_TEST_FACTS_LIST}} [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_180_RINGING_FAILS|180_RINGING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_BASIC_CALL_FAILS|BASIC_CALL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLIR_FAILS|CLIR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_ONNET_FAILS|CLNS_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CLNS_FAILS|CLNS]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_DIFF_FAILS|CONN_NR_DIFF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_INCOMING_FAILS|CONN_NR_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_CONN_NR_FAILS|CONN_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_DTMF_FAILS|DTMF]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_EARLY_MEDIA_INBOUND_FAILS|EARLY_MEDIA_INBOUND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_AUDIO_FAILS|FAX_AUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_ONNET_FAILS|FAX_T38_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38_FAILS|FAX_T38]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_FAX_T38ANDAUDIO_FAILS|FAX_T38ANDAUDIO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_ONNET_FAILS|G711A_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711A_FAILS|G711A]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_ONNET_FAILS|G711U_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G711U_FAILS|G711U]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_ONNET_FAILS|G722_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G722_FAILS|G722]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_ONNET_FAILS|G729_ONNET]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_G729_FAILS|G729]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_HOLD_RETRIEVE_FAILS|HOLD_RETRIEVE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_IP_FRAGMENTATION_FAILS|IP_FRAGMENTATION]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_LARGE_SIP_MESSAGES_FAILS|LARGE_SIP_MESSAGES]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_MOBILITY_FAILS|MOBILITY]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_NB_FAILS|OPUS_NB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_OPUS_WB_FAILS|OPUS_WB]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_RALERT_DISC_FAILS|RALERT_DISC]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_302_FAILS|REDIR_302]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_DIVHDR_FAILS|REDIR_DIVHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REDIR_HISTHDR_FAILS|REDIR_HISTHDR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_REVERSE_MEDIA_FAILS|REVERSE_MEDIA]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_ICE_FAILS|SDP_ICE]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_RTCP_MUX_FAILS|SDP_RTCP_MUX]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SDP_VIDEO_FAILS|SDP_VIDEO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SIP_INFO_FAILS|SIP_INFO]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INCOMING_FAILS|SRTP_INCOMING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_INTERNAL_FAILS|SRTP_INTERNAL]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SRTP_OUTGOING_FAILS|SRTP_OUTGOING]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_SUBSCRIBER_NR_FAILS|SUBSCRIBER_NR]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_BLIND_FAILS|XFER_BLIND]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_ALERT_FAILS|XFER_CONS_ALERT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_EXT_FAILS|XFER_CONS_EXT]], [[Template:SIP_TEST_FACT_DESCRIPTION_TEST_XFER_CONS_FAILS|XFER_CONS]]

== Test Results ==
{{SIP_TEST_TESTRESULT_ONLYMR_INTRO}}
=== {{SIP_TEST_RESULTS_MR_TITLE}} ===
; Registration : {{Template:SIP_Profile_Test_Registration_TCP_no_UDP}}

; NAT Traversal : {{Template:SIP_Profile_Test_NAT_a_no_c}}

; DTMF (RFC2833) : {{Template:SIP_Profile_Test_DTMF_RFC2833_yes}}

; Session Timer : {{Template:SIP_Profile_Test_EXPIRES_yes}}

; Redundancy : {{Template:SIP_Profile_Test_REDUNDANCY_no}}

; Correct signalling of Ringing-state : {{Template:SIP_Profile_Test_RINGING_yes}}

; CLIR : {{Template:SIP_Profile_Test_CLIR_yes}}

; Clip No Screening (CLNS) : {{Template:SIP_Profile_Test_CLNS_yes}} {{Template:SIP_Profile_Test_CLNS_clns_302_not_recommended}}

; COLP : {{Template:SIP_Profile_Test_COLP_out_yes_in_yes}} {{Template:SIP_Profile_Test_COLP_diff_no}}

; Early-Media : {{Template:SIP_Profile_Test_EARLY_MEDIA_INBOUND_no}}

; Fax : {{Template:SIP_Profile_Test_AUDIOFAX_PSTN_yes}}
: {{Template:SIP_Profile_Test_T38_PSTN_no_onnet_yes_fallback_no}}

; Codecs : supported to/from PSTN: G711A
: supported onnet (VoIP to VoIP): G711A, G711U and G722

; IP-Fragmentation : {{Template:SIP_Profile_Test_FRAGMENTATION_yes}}

; Large SIP messages : {{Template:SIP_Profile_Test_LARGE_MESSAGES_yes}}

; Reverse Media Negotiation : {{Template:SIP_Profile_Test_REV_MEDIA_NEG_yes}}

; Mobility Calls : {{Template:SIP_Profile_Test_MobilityCall_no_with_MediaRelay}}

; SRTP : {{Template:SIP_Profile_Test_SRTP_no}}

; Dialing of Subscriber Numbers : {{Template:SIP_Profile_Test_SUBSCRIBER_NR_yes}}

; Call Transfer : {{Template:SIP_Profile_Test_CALL_TRANSFER_ok}}

==Configuration==
Use profile ''DE-Vodafone-IP_Anlagen_Anschluss_TCP'' in ''Gateway/Interfaces/SIP'' to configure this SIP provider.
: {{SIP_TEST_V12_HINT}}

== Disclaimer ==
{{SIP_TEST_PREFACE}}

[[Category:Compat|{{PAGENAME}}]]
[[Category:3rdParty SIP Provider|{{PAGENAME}}]]